Introduction of VOIP

Definitions

bull Internet Telephony In the beginning Internet telephony simply meant the technology and techniques to let you make voice phone calls ndashlocal long distance and international ndashover the Internet using your PChellipthe definition of Internet telephony is broadening day by day to include all forms of media (voice video image) and all forms of messaging and all variations of speed from real-time to time-delayedrdquo

bull IP Telephony As defined by Microsoft) ldquoIP Telephony is an emerging set of technologies that enables voice data and video collaboration over existing IP-based LANs WANs and the Internet Specifically IP Telephony uses open IETF and ITU standards to move multimedia traffic over any network that uses IP (the Internet Protocol)rdquo

bull Voice over IP (VoIP) ldquoThe technology used to transmit voice conversations over a data network using the Internet Protocol Such data network may be the Internet or a corporate Intranet or managed networks typically used by long and local service traditional providers and ISPs that use VoIPrdquo

What is VOIP

bull VoIP is a term used in IP telephony to describe a set of facilities for managing the delivery of voice information using the Internet Protocol

mdash This means sending voice information in digital form in discrete packets rather than in the circuit committed protocols of the Public Switched Telephone Network (PSTN)

Understanding VOIP

VoIP stands for ldquoVoice over the Internet Protocolrdquo and is also referred to as IP telephony This technology enables users to

Make calls by moving packets of information over the Internet using broadband network connection to make phone calls to other VoIP and regular phones users rather than traditional public switched telephony circuits also known as rdquoPublic Switched Telephone Network (PSTN)rdquo or analog telephones

VOIP Overview

bull IP Telephony is a process that enables the transfer of voice data over a packet switched network as opposed to the traditional circuit switched network

bull The transmission of voice packets over the Internet Protocol (IP) is known as VoIP and is implemented properly holds the promise of converged networks and unified communications

VOIP NetworkMajor components of the VOIP network include

bull Signaling Protocols and Standards Initiate and control communications of voice video and data

- H323 Network platform and application independent standard that allows interoperability between H323-compliant devices - SIP Protocol that is based upon request-response or INVITE model SIP is used to establish conferencing telephony multimedia and other Internet communication sessions SIP uses a media-description language such as Hypertext Transfer Protocol (HTTP)

bull Media Gateway - Converts one media stream to another for example converts voice packets to analog - Can interact with call controllers proxies and soft switches via proprietary or standard protocols such as SIP

bull H323 Gateway - Transforms audio received from a telephone device or telecommunications system into a format that the data network can use - Acts as a bridge to the IP network - Generally has built-in intelligence to select the voice compression Codecs and adjust the protocols and timing between two dissimilar computer systems or voice over data networks

bull Call Server - Receives call setup request messages - Determines the status of destination devices and checks the authorization of users to originate or receive calls - Creates and sends the necessary messages to process the call requests

bull Gatekeeper - Provides call control media access and bandwidth management between endpoints - Performs address translation admissions control bandwidth management and zone control - Coordinates access to other servers and manages call routing - Receives requests from clients determines the destination server that it needs to communicate with and coordinates access with that server - Maps destination telephone number to destination endpoint IP address

bull IP Terminals and Clients - Include end points on the network such as hard or soft telephones (portable or stationary) and wireless devices (80211a and 80211b) - Bring voice and data communications to the end user A Call Server completes the call processingbull IP Backbone Network - Provides the universal communication language and foundation to allow dissimilar networks and equipment from a variety of vendors to interconnect

H3231048708 Recommendation published by ITU1048708 Ties together a number of protocols to allow multimedia transmission through an unreliable packet-based network1048708 1996 approved by ITU1048708 2003 Version 5

H323 Architecture

H323 Protocol Stack for VoIP

H323 Protocol SuiteVideo Audio Data TransportH261H263

G711G722

G7231G728G729

T122T124T125T126T127

H225H235H245

H4501H4502 H4503

RTPX2240

G7xx ndash Speech (De)Coding1048708 H323 systems must support G711 PCM 64kbps1048708 Other codecs G729 G726 hellip

RTP1048708 Realtime Transport Protocol(RFC 3550 July 2003)1048708 Application layer protocol for transmitting realtimedata (audio video )1048708 Includes payload type identification sequencenumbering timestamping delivery monitoring1048708 Mostly over UDP1048708 Supports multicast amp unicast

Control Protocol - RTCP1048708 RTP Control Protocol (RFC 3550 July 2003)1048708 Periodic transmission of control packets to all participants in the session1048708 Main functions- provide feedback on quality of data distribution- carries a persistent transport-level identifier for an RTP source (CNAME)- each participant sends control packets to all others which independently observe the number of participants

More Control Protocols in H3231048708 H225 (RAS)

- protocol between terminal and gatekeeper (if present)- allows terminals to joinleave zone requestreturnbandwidth provide status updates hellip

1048708 H245 (Call Control)- Media Control Protocol- Allows terminals to negotiate connection parameters (codec bit rate )

1048708 Q931 (Call Signalling)- Manages call setup and termination

SIP ndash Session Initiation Protocol1048708 Developed by IETF since 19991048708 RFC 2543 March 1999 (obsolete)1048708 RFC 3261 June 20021048708 Target develop simpler and more modular protocol for VoIP than the large and complex H323 by ITU1048708SIP is a text-based protocol similar to HTTP and SMTP for initiating interactive communication sessions between users1048708 SIP is an application-layer control (signalling) protocol for creating modifying and terminating sessions with one or more participants1048708 Sessions include Internet Multimedia conferences Internet Telephone calls and Multimedia distribution

SIP (2)1048708 SIP can be used with different transport protocols it doesnt even require reliable transport protocols1048708 A simple SIP client can be implemented using only UDP

SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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Definitions

bull Internet Telephony In the beginning Internet telephony simply meant the technology and techniques to let you make voice phone calls ndashlocal long distance and international ndashover the Internet using your PChellipthe definition of Internet telephony is broadening day by day to include all forms of media (voice video image) and all forms of messaging and all variations of speed from real-time to time-delayedrdquo

bull IP Telephony As defined by Microsoft) ldquoIP Telephony is an emerging set of technologies that enables voice data and video collaboration over existing IP-based LANs WANs and the Internet Specifically IP Telephony uses open IETF and ITU standards to move multimedia traffic over any network that uses IP (the Internet Protocol)rdquo

bull Voice over IP (VoIP) ldquoThe technology used to transmit voice conversations over a data network using the Internet Protocol Such data network may be the Internet or a corporate Intranet or managed networks typically used by long and local service traditional providers and ISPs that use VoIPrdquo

What is VOIP

bull VoIP is a term used in IP telephony to describe a set of facilities for managing the delivery of voice information using the Internet Protocol

mdash This means sending voice information in digital form in discrete packets rather than in the circuit committed protocols of the Public Switched Telephone Network (PSTN)

Understanding VOIP

VoIP stands for ldquoVoice over the Internet Protocolrdquo and is also referred to as IP telephony This technology enables users to

Make calls by moving packets of information over the Internet using broadband network connection to make phone calls to other VoIP and regular phones users rather than traditional public switched telephony circuits also known as rdquoPublic Switched Telephone Network (PSTN)rdquo or analog telephones

VOIP Overview

bull IP Telephony is a process that enables the transfer of voice data over a packet switched network as opposed to the traditional circuit switched network

bull The transmission of voice packets over the Internet Protocol (IP) is known as VoIP and is implemented properly holds the promise of converged networks and unified communications

VOIP NetworkMajor components of the VOIP network include

bull Signaling Protocols and Standards Initiate and control communications of voice video and data

- H323 Network platform and application independent standard that allows interoperability between H323-compliant devices - SIP Protocol that is based upon request-response or INVITE model SIP is used to establish conferencing telephony multimedia and other Internet communication sessions SIP uses a media-description language such as Hypertext Transfer Protocol (HTTP)

bull Media Gateway - Converts one media stream to another for example converts voice packets to analog - Can interact with call controllers proxies and soft switches via proprietary or standard protocols such as SIP

bull H323 Gateway - Transforms audio received from a telephone device or telecommunications system into a format that the data network can use - Acts as a bridge to the IP network - Generally has built-in intelligence to select the voice compression Codecs and adjust the protocols and timing between two dissimilar computer systems or voice over data networks

bull Call Server - Receives call setup request messages - Determines the status of destination devices and checks the authorization of users to originate or receive calls - Creates and sends the necessary messages to process the call requests

bull Gatekeeper - Provides call control media access and bandwidth management between endpoints - Performs address translation admissions control bandwidth management and zone control - Coordinates access to other servers and manages call routing - Receives requests from clients determines the destination server that it needs to communicate with and coordinates access with that server - Maps destination telephone number to destination endpoint IP address

bull IP Terminals and Clients - Include end points on the network such as hard or soft telephones (portable or stationary) and wireless devices (80211a and 80211b) - Bring voice and data communications to the end user A Call Server completes the call processingbull IP Backbone Network - Provides the universal communication language and foundation to allow dissimilar networks and equipment from a variety of vendors to interconnect

H3231048708 Recommendation published by ITU1048708 Ties together a number of protocols to allow multimedia transmission through an unreliable packet-based network1048708 1996 approved by ITU1048708 2003 Version 5

H323 Architecture

H323 Protocol Stack for VoIP

H323 Protocol SuiteVideo Audio Data TransportH261H263

G711G722

G7231G728G729

T122T124T125T126T127

H225H235H245

H4501H4502 H4503

RTPX2240

G7xx ndash Speech (De)Coding1048708 H323 systems must support G711 PCM 64kbps1048708 Other codecs G729 G726 hellip

RTP1048708 Realtime Transport Protocol(RFC 3550 July 2003)1048708 Application layer protocol for transmitting realtimedata (audio video )1048708 Includes payload type identification sequencenumbering timestamping delivery monitoring1048708 Mostly over UDP1048708 Supports multicast amp unicast

Control Protocol - RTCP1048708 RTP Control Protocol (RFC 3550 July 2003)1048708 Periodic transmission of control packets to all participants in the session1048708 Main functions- provide feedback on quality of data distribution- carries a persistent transport-level identifier for an RTP source (CNAME)- each participant sends control packets to all others which independently observe the number of participants

More Control Protocols in H3231048708 H225 (RAS)

- protocol between terminal and gatekeeper (if present)- allows terminals to joinleave zone requestreturnbandwidth provide status updates hellip

1048708 H245 (Call Control)- Media Control Protocol- Allows terminals to negotiate connection parameters (codec bit rate )

1048708 Q931 (Call Signalling)- Manages call setup and termination

SIP ndash Session Initiation Protocol1048708 Developed by IETF since 19991048708 RFC 2543 March 1999 (obsolete)1048708 RFC 3261 June 20021048708 Target develop simpler and more modular protocol for VoIP than the large and complex H323 by ITU1048708SIP is a text-based protocol similar to HTTP and SMTP for initiating interactive communication sessions between users1048708 SIP is an application-layer control (signalling) protocol for creating modifying and terminating sessions with one or more participants1048708 Sessions include Internet Multimedia conferences Internet Telephone calls and Multimedia distribution

SIP (2)1048708 SIP can be used with different transport protocols it doesnt even require reliable transport protocols1048708 A simple SIP client can be implemented using only UDP

SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

• Slide 1
• Slide 2
• Slide 3
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• Slide 26

What is VOIP

bull VoIP is a term used in IP telephony to describe a set of facilities for managing the delivery of voice information using the Internet Protocol

mdash This means sending voice information in digital form in discrete packets rather than in the circuit committed protocols of the Public Switched Telephone Network (PSTN)

Understanding VOIP

VoIP stands for ldquoVoice over the Internet Protocolrdquo and is also referred to as IP telephony This technology enables users to

Make calls by moving packets of information over the Internet using broadband network connection to make phone calls to other VoIP and regular phones users rather than traditional public switched telephony circuits also known as rdquoPublic Switched Telephone Network (PSTN)rdquo or analog telephones

VOIP Overview

bull IP Telephony is a process that enables the transfer of voice data over a packet switched network as opposed to the traditional circuit switched network

bull The transmission of voice packets over the Internet Protocol (IP) is known as VoIP and is implemented properly holds the promise of converged networks and unified communications

VOIP NetworkMajor components of the VOIP network include

bull Signaling Protocols and Standards Initiate and control communications of voice video and data

- H323 Network platform and application independent standard that allows interoperability between H323-compliant devices - SIP Protocol that is based upon request-response or INVITE model SIP is used to establish conferencing telephony multimedia and other Internet communication sessions SIP uses a media-description language such as Hypertext Transfer Protocol (HTTP)

bull Media Gateway - Converts one media stream to another for example converts voice packets to analog - Can interact with call controllers proxies and soft switches via proprietary or standard protocols such as SIP

bull H323 Gateway - Transforms audio received from a telephone device or telecommunications system into a format that the data network can use - Acts as a bridge to the IP network - Generally has built-in intelligence to select the voice compression Codecs and adjust the protocols and timing between two dissimilar computer systems or voice over data networks

bull Call Server - Receives call setup request messages - Determines the status of destination devices and checks the authorization of users to originate or receive calls - Creates and sends the necessary messages to process the call requests

bull Gatekeeper - Provides call control media access and bandwidth management between endpoints - Performs address translation admissions control bandwidth management and zone control - Coordinates access to other servers and manages call routing - Receives requests from clients determines the destination server that it needs to communicate with and coordinates access with that server - Maps destination telephone number to destination endpoint IP address

bull IP Terminals and Clients - Include end points on the network such as hard or soft telephones (portable or stationary) and wireless devices (80211a and 80211b) - Bring voice and data communications to the end user A Call Server completes the call processingbull IP Backbone Network - Provides the universal communication language and foundation to allow dissimilar networks and equipment from a variety of vendors to interconnect

H3231048708 Recommendation published by ITU1048708 Ties together a number of protocols to allow multimedia transmission through an unreliable packet-based network1048708 1996 approved by ITU1048708 2003 Version 5

H323 Architecture

H323 Protocol Stack for VoIP

H323 Protocol SuiteVideo Audio Data TransportH261H263

G711G722

G7231G728G729

T122T124T125T126T127

H225H235H245

H4501H4502 H4503

RTPX2240

G7xx ndash Speech (De)Coding1048708 H323 systems must support G711 PCM 64kbps1048708 Other codecs G729 G726 hellip

RTP1048708 Realtime Transport Protocol(RFC 3550 July 2003)1048708 Application layer protocol for transmitting realtimedata (audio video )1048708 Includes payload type identification sequencenumbering timestamping delivery monitoring1048708 Mostly over UDP1048708 Supports multicast amp unicast

Control Protocol - RTCP1048708 RTP Control Protocol (RFC 3550 July 2003)1048708 Periodic transmission of control packets to all participants in the session1048708 Main functions- provide feedback on quality of data distribution- carries a persistent transport-level identifier for an RTP source (CNAME)- each participant sends control packets to all others which independently observe the number of participants

More Control Protocols in H3231048708 H225 (RAS)

- protocol between terminal and gatekeeper (if present)- allows terminals to joinleave zone requestreturnbandwidth provide status updates hellip

1048708 H245 (Call Control)- Media Control Protocol- Allows terminals to negotiate connection parameters (codec bit rate )

1048708 Q931 (Call Signalling)- Manages call setup and termination

SIP ndash Session Initiation Protocol1048708 Developed by IETF since 19991048708 RFC 2543 March 1999 (obsolete)1048708 RFC 3261 June 20021048708 Target develop simpler and more modular protocol for VoIP than the large and complex H323 by ITU1048708SIP is a text-based protocol similar to HTTP and SMTP for initiating interactive communication sessions between users1048708 SIP is an application-layer control (signalling) protocol for creating modifying and terminating sessions with one or more participants1048708 Sessions include Internet Multimedia conferences Internet Telephone calls and Multimedia distribution

SIP (2)1048708 SIP can be used with different transport protocols it doesnt even require reliable transport protocols1048708 A simple SIP client can be implemented using only UDP

SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

• Slide 1
• Slide 2
• Slide 3
• Slide 4
• Slide 5
• Slide 6
• Slide 7
• Slide 8
• Slide 9
• Slide 10
• Slide 11
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• Slide 13
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• Slide 17
• Slide 18
• Slide 19
• Slide 20
• Slide 21
• Slide 22
• Slide 23
• Slide 24
• Slide 25
• Slide 26

Understanding VOIP

VoIP stands for ldquoVoice over the Internet Protocolrdquo and is also referred to as IP telephony This technology enables users to

Make calls by moving packets of information over the Internet using broadband network connection to make phone calls to other VoIP and regular phones users rather than traditional public switched telephony circuits also known as rdquoPublic Switched Telephone Network (PSTN)rdquo or analog telephones

VOIP Overview

bull IP Telephony is a process that enables the transfer of voice data over a packet switched network as opposed to the traditional circuit switched network

bull The transmission of voice packets over the Internet Protocol (IP) is known as VoIP and is implemented properly holds the promise of converged networks and unified communications

VOIP NetworkMajor components of the VOIP network include

bull Signaling Protocols and Standards Initiate and control communications of voice video and data

- H323 Network platform and application independent standard that allows interoperability between H323-compliant devices - SIP Protocol that is based upon request-response or INVITE model SIP is used to establish conferencing telephony multimedia and other Internet communication sessions SIP uses a media-description language such as Hypertext Transfer Protocol (HTTP)

bull Media Gateway - Converts one media stream to another for example converts voice packets to analog - Can interact with call controllers proxies and soft switches via proprietary or standard protocols such as SIP

bull H323 Gateway - Transforms audio received from a telephone device or telecommunications system into a format that the data network can use - Acts as a bridge to the IP network - Generally has built-in intelligence to select the voice compression Codecs and adjust the protocols and timing between two dissimilar computer systems or voice over data networks

bull Call Server - Receives call setup request messages - Determines the status of destination devices and checks the authorization of users to originate or receive calls - Creates and sends the necessary messages to process the call requests

bull Gatekeeper - Provides call control media access and bandwidth management between endpoints - Performs address translation admissions control bandwidth management and zone control - Coordinates access to other servers and manages call routing - Receives requests from clients determines the destination server that it needs to communicate with and coordinates access with that server - Maps destination telephone number to destination endpoint IP address

bull IP Terminals and Clients - Include end points on the network such as hard or soft telephones (portable or stationary) and wireless devices (80211a and 80211b) - Bring voice and data communications to the end user A Call Server completes the call processingbull IP Backbone Network - Provides the universal communication language and foundation to allow dissimilar networks and equipment from a variety of vendors to interconnect

H3231048708 Recommendation published by ITU1048708 Ties together a number of protocols to allow multimedia transmission through an unreliable packet-based network1048708 1996 approved by ITU1048708 2003 Version 5

H323 Architecture

H323 Protocol Stack for VoIP

H323 Protocol SuiteVideo Audio Data TransportH261H263

G711G722

G7231G728G729

T122T124T125T126T127

H225H235H245

H4501H4502 H4503

RTPX2240

G7xx ndash Speech (De)Coding1048708 H323 systems must support G711 PCM 64kbps1048708 Other codecs G729 G726 hellip

RTP1048708 Realtime Transport Protocol(RFC 3550 July 2003)1048708 Application layer protocol for transmitting realtimedata (audio video )1048708 Includes payload type identification sequencenumbering timestamping delivery monitoring1048708 Mostly over UDP1048708 Supports multicast amp unicast

Control Protocol - RTCP1048708 RTP Control Protocol (RFC 3550 July 2003)1048708 Periodic transmission of control packets to all participants in the session1048708 Main functions- provide feedback on quality of data distribution- carries a persistent transport-level identifier for an RTP source (CNAME)- each participant sends control packets to all others which independently observe the number of participants

More Control Protocols in H3231048708 H225 (RAS)

- protocol between terminal and gatekeeper (if present)- allows terminals to joinleave zone requestreturnbandwidth provide status updates hellip

1048708 H245 (Call Control)- Media Control Protocol- Allows terminals to negotiate connection parameters (codec bit rate )

1048708 Q931 (Call Signalling)- Manages call setup and termination

SIP ndash Session Initiation Protocol1048708 Developed by IETF since 19991048708 RFC 2543 March 1999 (obsolete)1048708 RFC 3261 June 20021048708 Target develop simpler and more modular protocol for VoIP than the large and complex H323 by ITU1048708SIP is a text-based protocol similar to HTTP and SMTP for initiating interactive communication sessions between users1048708 SIP is an application-layer control (signalling) protocol for creating modifying and terminating sessions with one or more participants1048708 Sessions include Internet Multimedia conferences Internet Telephone calls and Multimedia distribution

SIP (2)1048708 SIP can be used with different transport protocols it doesnt even require reliable transport protocols1048708 A simple SIP client can be implemented using only UDP

SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

• Slide 1
• Slide 2
• Slide 3
• Slide 4
• Slide 5
• Slide 6
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VOIP Overview

bull IP Telephony is a process that enables the transfer of voice data over a packet switched network as opposed to the traditional circuit switched network

bull The transmission of voice packets over the Internet Protocol (IP) is known as VoIP and is implemented properly holds the promise of converged networks and unified communications

VOIP NetworkMajor components of the VOIP network include

bull Signaling Protocols and Standards Initiate and control communications of voice video and data

- H323 Network platform and application independent standard that allows interoperability between H323-compliant devices - SIP Protocol that is based upon request-response or INVITE model SIP is used to establish conferencing telephony multimedia and other Internet communication sessions SIP uses a media-description language such as Hypertext Transfer Protocol (HTTP)

bull Media Gateway - Converts one media stream to another for example converts voice packets to analog - Can interact with call controllers proxies and soft switches via proprietary or standard protocols such as SIP

bull H323 Gateway - Transforms audio received from a telephone device or telecommunications system into a format that the data network can use - Acts as a bridge to the IP network - Generally has built-in intelligence to select the voice compression Codecs and adjust the protocols and timing between two dissimilar computer systems or voice over data networks

bull Call Server - Receives call setup request messages - Determines the status of destination devices and checks the authorization of users to originate or receive calls - Creates and sends the necessary messages to process the call requests

bull Gatekeeper - Provides call control media access and bandwidth management between endpoints - Performs address translation admissions control bandwidth management and zone control - Coordinates access to other servers and manages call routing - Receives requests from clients determines the destination server that it needs to communicate with and coordinates access with that server - Maps destination telephone number to destination endpoint IP address

bull IP Terminals and Clients - Include end points on the network such as hard or soft telephones (portable or stationary) and wireless devices (80211a and 80211b) - Bring voice and data communications to the end user A Call Server completes the call processingbull IP Backbone Network - Provides the universal communication language and foundation to allow dissimilar networks and equipment from a variety of vendors to interconnect

H3231048708 Recommendation published by ITU1048708 Ties together a number of protocols to allow multimedia transmission through an unreliable packet-based network1048708 1996 approved by ITU1048708 2003 Version 5

H323 Architecture

H323 Protocol Stack for VoIP

H323 Protocol SuiteVideo Audio Data TransportH261H263

G711G722

G7231G728G729

T122T124T125T126T127

H225H235H245

H4501H4502 H4503

RTPX2240

G7xx ndash Speech (De)Coding1048708 H323 systems must support G711 PCM 64kbps1048708 Other codecs G729 G726 hellip

RTP1048708 Realtime Transport Protocol(RFC 3550 July 2003)1048708 Application layer protocol for transmitting realtimedata (audio video )1048708 Includes payload type identification sequencenumbering timestamping delivery monitoring1048708 Mostly over UDP1048708 Supports multicast amp unicast

Control Protocol - RTCP1048708 RTP Control Protocol (RFC 3550 July 2003)1048708 Periodic transmission of control packets to all participants in the session1048708 Main functions- provide feedback on quality of data distribution- carries a persistent transport-level identifier for an RTP source (CNAME)- each participant sends control packets to all others which independently observe the number of participants

More Control Protocols in H3231048708 H225 (RAS)

- protocol between terminal and gatekeeper (if present)- allows terminals to joinleave zone requestreturnbandwidth provide status updates hellip

1048708 H245 (Call Control)- Media Control Protocol- Allows terminals to negotiate connection parameters (codec bit rate )

1048708 Q931 (Call Signalling)- Manages call setup and termination

SIP ndash Session Initiation Protocol1048708 Developed by IETF since 19991048708 RFC 2543 March 1999 (obsolete)1048708 RFC 3261 June 20021048708 Target develop simpler and more modular protocol for VoIP than the large and complex H323 by ITU1048708SIP is a text-based protocol similar to HTTP and SMTP for initiating interactive communication sessions between users1048708 SIP is an application-layer control (signalling) protocol for creating modifying and terminating sessions with one or more participants1048708 Sessions include Internet Multimedia conferences Internet Telephone calls and Multimedia distribution

SIP (2)1048708 SIP can be used with different transport protocols it doesnt even require reliable transport protocols1048708 A simple SIP client can be implemented using only UDP

SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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VOIP NetworkMajor components of the VOIP network include

bull Signaling Protocols and Standards Initiate and control communications of voice video and data

- H323 Network platform and application independent standard that allows interoperability between H323-compliant devices - SIP Protocol that is based upon request-response or INVITE model SIP is used to establish conferencing telephony multimedia and other Internet communication sessions SIP uses a media-description language such as Hypertext Transfer Protocol (HTTP)

bull Media Gateway - Converts one media stream to another for example converts voice packets to analog - Can interact with call controllers proxies and soft switches via proprietary or standard protocols such as SIP

bull H323 Gateway - Transforms audio received from a telephone device or telecommunications system into a format that the data network can use - Acts as a bridge to the IP network - Generally has built-in intelligence to select the voice compression Codecs and adjust the protocols and timing between two dissimilar computer systems or voice over data networks

bull Call Server - Receives call setup request messages - Determines the status of destination devices and checks the authorization of users to originate or receive calls - Creates and sends the necessary messages to process the call requests

bull Gatekeeper - Provides call control media access and bandwidth management between endpoints - Performs address translation admissions control bandwidth management and zone control - Coordinates access to other servers and manages call routing - Receives requests from clients determines the destination server that it needs to communicate with and coordinates access with that server - Maps destination telephone number to destination endpoint IP address

bull IP Terminals and Clients - Include end points on the network such as hard or soft telephones (portable or stationary) and wireless devices (80211a and 80211b) - Bring voice and data communications to the end user A Call Server completes the call processingbull IP Backbone Network - Provides the universal communication language and foundation to allow dissimilar networks and equipment from a variety of vendors to interconnect

H3231048708 Recommendation published by ITU1048708 Ties together a number of protocols to allow multimedia transmission through an unreliable packet-based network1048708 1996 approved by ITU1048708 2003 Version 5

H323 Architecture

H323 Protocol Stack for VoIP

H323 Protocol SuiteVideo Audio Data TransportH261H263

G711G722

G7231G728G729

T122T124T125T126T127

H225H235H245

H4501H4502 H4503

RTPX2240

G7xx ndash Speech (De)Coding1048708 H323 systems must support G711 PCM 64kbps1048708 Other codecs G729 G726 hellip

RTP1048708 Realtime Transport Protocol(RFC 3550 July 2003)1048708 Application layer protocol for transmitting realtimedata (audio video )1048708 Includes payload type identification sequencenumbering timestamping delivery monitoring1048708 Mostly over UDP1048708 Supports multicast amp unicast

Control Protocol - RTCP1048708 RTP Control Protocol (RFC 3550 July 2003)1048708 Periodic transmission of control packets to all participants in the session1048708 Main functions- provide feedback on quality of data distribution- carries a persistent transport-level identifier for an RTP source (CNAME)- each participant sends control packets to all others which independently observe the number of participants

More Control Protocols in H3231048708 H225 (RAS)

- protocol between terminal and gatekeeper (if present)- allows terminals to joinleave zone requestreturnbandwidth provide status updates hellip

1048708 H245 (Call Control)- Media Control Protocol- Allows terminals to negotiate connection parameters (codec bit rate )

1048708 Q931 (Call Signalling)- Manages call setup and termination

SIP ndash Session Initiation Protocol1048708 Developed by IETF since 19991048708 RFC 2543 March 1999 (obsolete)1048708 RFC 3261 June 20021048708 Target develop simpler and more modular protocol for VoIP than the large and complex H323 by ITU1048708SIP is a text-based protocol similar to HTTP and SMTP for initiating interactive communication sessions between users1048708 SIP is an application-layer control (signalling) protocol for creating modifying and terminating sessions with one or more participants1048708 Sessions include Internet Multimedia conferences Internet Telephone calls and Multimedia distribution

SIP (2)1048708 SIP can be used with different transport protocols it doesnt even require reliable transport protocols1048708 A simple SIP client can be implemented using only UDP

SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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bull H323 Gateway - Transforms audio received from a telephone device or telecommunications system into a format that the data network can use - Acts as a bridge to the IP network - Generally has built-in intelligence to select the voice compression Codecs and adjust the protocols and timing between two dissimilar computer systems or voice over data networks

bull Call Server - Receives call setup request messages - Determines the status of destination devices and checks the authorization of users to originate or receive calls - Creates and sends the necessary messages to process the call requests

bull Gatekeeper - Provides call control media access and bandwidth management between endpoints - Performs address translation admissions control bandwidth management and zone control - Coordinates access to other servers and manages call routing - Receives requests from clients determines the destination server that it needs to communicate with and coordinates access with that server - Maps destination telephone number to destination endpoint IP address

bull IP Terminals and Clients - Include end points on the network such as hard or soft telephones (portable or stationary) and wireless devices (80211a and 80211b) - Bring voice and data communications to the end user A Call Server completes the call processingbull IP Backbone Network - Provides the universal communication language and foundation to allow dissimilar networks and equipment from a variety of vendors to interconnect

H3231048708 Recommendation published by ITU1048708 Ties together a number of protocols to allow multimedia transmission through an unreliable packet-based network1048708 1996 approved by ITU1048708 2003 Version 5

H323 Architecture

H323 Protocol Stack for VoIP

H323 Protocol SuiteVideo Audio Data TransportH261H263

G711G722

G7231G728G729

T122T124T125T126T127

H225H235H245

H4501H4502 H4503

RTPX2240

G7xx ndash Speech (De)Coding1048708 H323 systems must support G711 PCM 64kbps1048708 Other codecs G729 G726 hellip

RTP1048708 Realtime Transport Protocol(RFC 3550 July 2003)1048708 Application layer protocol for transmitting realtimedata (audio video )1048708 Includes payload type identification sequencenumbering timestamping delivery monitoring1048708 Mostly over UDP1048708 Supports multicast amp unicast

Control Protocol - RTCP1048708 RTP Control Protocol (RFC 3550 July 2003)1048708 Periodic transmission of control packets to all participants in the session1048708 Main functions- provide feedback on quality of data distribution- carries a persistent transport-level identifier for an RTP source (CNAME)- each participant sends control packets to all others which independently observe the number of participants

More Control Protocols in H3231048708 H225 (RAS)

- protocol between terminal and gatekeeper (if present)- allows terminals to joinleave zone requestreturnbandwidth provide status updates hellip

1048708 H245 (Call Control)- Media Control Protocol- Allows terminals to negotiate connection parameters (codec bit rate )

1048708 Q931 (Call Signalling)- Manages call setup and termination

SIP ndash Session Initiation Protocol1048708 Developed by IETF since 19991048708 RFC 2543 March 1999 (obsolete)1048708 RFC 3261 June 20021048708 Target develop simpler and more modular protocol for VoIP than the large and complex H323 by ITU1048708SIP is a text-based protocol similar to HTTP and SMTP for initiating interactive communication sessions between users1048708 SIP is an application-layer control (signalling) protocol for creating modifying and terminating sessions with one or more participants1048708 Sessions include Internet Multimedia conferences Internet Telephone calls and Multimedia distribution

SIP (2)1048708 SIP can be used with different transport protocols it doesnt even require reliable transport protocols1048708 A simple SIP client can be implemented using only UDP

SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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bull Gatekeeper - Provides call control media access and bandwidth management between endpoints - Performs address translation admissions control bandwidth management and zone control - Coordinates access to other servers and manages call routing - Receives requests from clients determines the destination server that it needs to communicate with and coordinates access with that server - Maps destination telephone number to destination endpoint IP address

bull IP Terminals and Clients - Include end points on the network such as hard or soft telephones (portable or stationary) and wireless devices (80211a and 80211b) - Bring voice and data communications to the end user A Call Server completes the call processingbull IP Backbone Network - Provides the universal communication language and foundation to allow dissimilar networks and equipment from a variety of vendors to interconnect

H3231048708 Recommendation published by ITU1048708 Ties together a number of protocols to allow multimedia transmission through an unreliable packet-based network1048708 1996 approved by ITU1048708 2003 Version 5

H323 Architecture

H323 Protocol Stack for VoIP

H323 Protocol SuiteVideo Audio Data TransportH261H263

G711G722

G7231G728G729

T122T124T125T126T127

H225H235H245

H4501H4502 H4503

RTPX2240

G7xx ndash Speech (De)Coding1048708 H323 systems must support G711 PCM 64kbps1048708 Other codecs G729 G726 hellip

RTP1048708 Realtime Transport Protocol(RFC 3550 July 2003)1048708 Application layer protocol for transmitting realtimedata (audio video )1048708 Includes payload type identification sequencenumbering timestamping delivery monitoring1048708 Mostly over UDP1048708 Supports multicast amp unicast

Control Protocol - RTCP1048708 RTP Control Protocol (RFC 3550 July 2003)1048708 Periodic transmission of control packets to all participants in the session1048708 Main functions- provide feedback on quality of data distribution- carries a persistent transport-level identifier for an RTP source (CNAME)- each participant sends control packets to all others which independently observe the number of participants

More Control Protocols in H3231048708 H225 (RAS)

- protocol between terminal and gatekeeper (if present)- allows terminals to joinleave zone requestreturnbandwidth provide status updates hellip

1048708 H245 (Call Control)- Media Control Protocol- Allows terminals to negotiate connection parameters (codec bit rate )

1048708 Q931 (Call Signalling)- Manages call setup and termination

SIP ndash Session Initiation Protocol1048708 Developed by IETF since 19991048708 RFC 2543 March 1999 (obsolete)1048708 RFC 3261 June 20021048708 Target develop simpler and more modular protocol for VoIP than the large and complex H323 by ITU1048708SIP is a text-based protocol similar to HTTP and SMTP for initiating interactive communication sessions between users1048708 SIP is an application-layer control (signalling) protocol for creating modifying and terminating sessions with one or more participants1048708 Sessions include Internet Multimedia conferences Internet Telephone calls and Multimedia distribution

SIP (2)1048708 SIP can be used with different transport protocols it doesnt even require reliable transport protocols1048708 A simple SIP client can be implemented using only UDP

SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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H3231048708 Recommendation published by ITU1048708 Ties together a number of protocols to allow multimedia transmission through an unreliable packet-based network1048708 1996 approved by ITU1048708 2003 Version 5

H323 Architecture

H323 Protocol Stack for VoIP

H323 Protocol SuiteVideo Audio Data TransportH261H263

G711G722

G7231G728G729

T122T124T125T126T127

H225H235H245

H4501H4502 H4503

RTPX2240

G7xx ndash Speech (De)Coding1048708 H323 systems must support G711 PCM 64kbps1048708 Other codecs G729 G726 hellip

RTP1048708 Realtime Transport Protocol(RFC 3550 July 2003)1048708 Application layer protocol for transmitting realtimedata (audio video )1048708 Includes payload type identification sequencenumbering timestamping delivery monitoring1048708 Mostly over UDP1048708 Supports multicast amp unicast

Control Protocol - RTCP1048708 RTP Control Protocol (RFC 3550 July 2003)1048708 Periodic transmission of control packets to all participants in the session1048708 Main functions- provide feedback on quality of data distribution- carries a persistent transport-level identifier for an RTP source (CNAME)- each participant sends control packets to all others which independently observe the number of participants

More Control Protocols in H3231048708 H225 (RAS)

- protocol between terminal and gatekeeper (if present)- allows terminals to joinleave zone requestreturnbandwidth provide status updates hellip

1048708 H245 (Call Control)- Media Control Protocol- Allows terminals to negotiate connection parameters (codec bit rate )

1048708 Q931 (Call Signalling)- Manages call setup and termination

SIP ndash Session Initiation Protocol1048708 Developed by IETF since 19991048708 RFC 2543 March 1999 (obsolete)1048708 RFC 3261 June 20021048708 Target develop simpler and more modular protocol for VoIP than the large and complex H323 by ITU1048708SIP is a text-based protocol similar to HTTP and SMTP for initiating interactive communication sessions between users1048708 SIP is an application-layer control (signalling) protocol for creating modifying and terminating sessions with one or more participants1048708 Sessions include Internet Multimedia conferences Internet Telephone calls and Multimedia distribution

SIP (2)1048708 SIP can be used with different transport protocols it doesnt even require reliable transport protocols1048708 A simple SIP client can be implemented using only UDP

SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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H323 Protocol Stack for VoIP

H323 Protocol SuiteVideo Audio Data TransportH261H263

G711G722

G7231G728G729

T122T124T125T126T127

H225H235H245

H4501H4502 H4503

RTPX2240

G7xx ndash Speech (De)Coding1048708 H323 systems must support G711 PCM 64kbps1048708 Other codecs G729 G726 hellip

RTP1048708 Realtime Transport Protocol(RFC 3550 July 2003)1048708 Application layer protocol for transmitting realtimedata (audio video )1048708 Includes payload type identification sequencenumbering timestamping delivery monitoring1048708 Mostly over UDP1048708 Supports multicast amp unicast

Control Protocol - RTCP1048708 RTP Control Protocol (RFC 3550 July 2003)1048708 Periodic transmission of control packets to all participants in the session1048708 Main functions- provide feedback on quality of data distribution- carries a persistent transport-level identifier for an RTP source (CNAME)- each participant sends control packets to all others which independently observe the number of participants

More Control Protocols in H3231048708 H225 (RAS)

- protocol between terminal and gatekeeper (if present)- allows terminals to joinleave zone requestreturnbandwidth provide status updates hellip

1048708 H245 (Call Control)- Media Control Protocol- Allows terminals to negotiate connection parameters (codec bit rate )

1048708 Q931 (Call Signalling)- Manages call setup and termination

SIP ndash Session Initiation Protocol1048708 Developed by IETF since 19991048708 RFC 2543 March 1999 (obsolete)1048708 RFC 3261 June 20021048708 Target develop simpler and more modular protocol for VoIP than the large and complex H323 by ITU1048708SIP is a text-based protocol similar to HTTP and SMTP for initiating interactive communication sessions between users1048708 SIP is an application-layer control (signalling) protocol for creating modifying and terminating sessions with one or more participants1048708 Sessions include Internet Multimedia conferences Internet Telephone calls and Multimedia distribution

SIP (2)1048708 SIP can be used with different transport protocols it doesnt even require reliable transport protocols1048708 A simple SIP client can be implemented using only UDP

SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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H323 Protocol SuiteVideo Audio Data TransportH261H263

G711G722

G7231G728G729

T122T124T125T126T127

H225H235H245

H4501H4502 H4503

RTPX2240

G7xx ndash Speech (De)Coding1048708 H323 systems must support G711 PCM 64kbps1048708 Other codecs G729 G726 hellip

RTP1048708 Realtime Transport Protocol(RFC 3550 July 2003)1048708 Application layer protocol for transmitting realtimedata (audio video )1048708 Includes payload type identification sequencenumbering timestamping delivery monitoring1048708 Mostly over UDP1048708 Supports multicast amp unicast

Control Protocol - RTCP1048708 RTP Control Protocol (RFC 3550 July 2003)1048708 Periodic transmission of control packets to all participants in the session1048708 Main functions- provide feedback on quality of data distribution- carries a persistent transport-level identifier for an RTP source (CNAME)- each participant sends control packets to all others which independently observe the number of participants

More Control Protocols in H3231048708 H225 (RAS)

- protocol between terminal and gatekeeper (if present)- allows terminals to joinleave zone requestreturnbandwidth provide status updates hellip

1048708 H245 (Call Control)- Media Control Protocol- Allows terminals to negotiate connection parameters (codec bit rate )

1048708 Q931 (Call Signalling)- Manages call setup and termination

SIP ndash Session Initiation Protocol1048708 Developed by IETF since 19991048708 RFC 2543 March 1999 (obsolete)1048708 RFC 3261 June 20021048708 Target develop simpler and more modular protocol for VoIP than the large and complex H323 by ITU1048708SIP is a text-based protocol similar to HTTP and SMTP for initiating interactive communication sessions between users1048708 SIP is an application-layer control (signalling) protocol for creating modifying and terminating sessions with one or more participants1048708 Sessions include Internet Multimedia conferences Internet Telephone calls and Multimedia distribution

SIP (2)1048708 SIP can be used with different transport protocols it doesnt even require reliable transport protocols1048708 A simple SIP client can be implemented using only UDP

SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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G7xx ndash Speech (De)Coding1048708 H323 systems must support G711 PCM 64kbps1048708 Other codecs G729 G726 hellip

RTP1048708 Realtime Transport Protocol(RFC 3550 July 2003)1048708 Application layer protocol for transmitting realtimedata (audio video )1048708 Includes payload type identification sequencenumbering timestamping delivery monitoring1048708 Mostly over UDP1048708 Supports multicast amp unicast

Control Protocol - RTCP1048708 RTP Control Protocol (RFC 3550 July 2003)1048708 Periodic transmission of control packets to all participants in the session1048708 Main functions- provide feedback on quality of data distribution- carries a persistent transport-level identifier for an RTP source (CNAME)- each participant sends control packets to all others which independently observe the number of participants

More Control Protocols in H3231048708 H225 (RAS)

- protocol between terminal and gatekeeper (if present)- allows terminals to joinleave zone requestreturnbandwidth provide status updates hellip

1048708 H245 (Call Control)- Media Control Protocol- Allows terminals to negotiate connection parameters (codec bit rate )

1048708 Q931 (Call Signalling)- Manages call setup and termination

SIP ndash Session Initiation Protocol1048708 Developed by IETF since 19991048708 RFC 2543 March 1999 (obsolete)1048708 RFC 3261 June 20021048708 Target develop simpler and more modular protocol for VoIP than the large and complex H323 by ITU1048708SIP is a text-based protocol similar to HTTP and SMTP for initiating interactive communication sessions between users1048708 SIP is an application-layer control (signalling) protocol for creating modifying and terminating sessions with one or more participants1048708 Sessions include Internet Multimedia conferences Internet Telephone calls and Multimedia distribution

SIP (2)1048708 SIP can be used with different transport protocols it doesnt even require reliable transport protocols1048708 A simple SIP client can be implemented using only UDP

SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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Control Protocol - RTCP1048708 RTP Control Protocol (RFC 3550 July 2003)1048708 Periodic transmission of control packets to all participants in the session1048708 Main functions- provide feedback on quality of data distribution- carries a persistent transport-level identifier for an RTP source (CNAME)- each participant sends control packets to all others which independently observe the number of participants

More Control Protocols in H3231048708 H225 (RAS)

- protocol between terminal and gatekeeper (if present)- allows terminals to joinleave zone requestreturnbandwidth provide status updates hellip

1048708 H245 (Call Control)- Media Control Protocol- Allows terminals to negotiate connection parameters (codec bit rate )

1048708 Q931 (Call Signalling)- Manages call setup and termination

SIP ndash Session Initiation Protocol1048708 Developed by IETF since 19991048708 RFC 2543 March 1999 (obsolete)1048708 RFC 3261 June 20021048708 Target develop simpler and more modular protocol for VoIP than the large and complex H323 by ITU1048708SIP is a text-based protocol similar to HTTP and SMTP for initiating interactive communication sessions between users1048708 SIP is an application-layer control (signalling) protocol for creating modifying and terminating sessions with one or more participants1048708 Sessions include Internet Multimedia conferences Internet Telephone calls and Multimedia distribution

SIP (2)1048708 SIP can be used with different transport protocols it doesnt even require reliable transport protocols1048708 A simple SIP client can be implemented using only UDP

SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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More Control Protocols in H3231048708 H225 (RAS)

- protocol between terminal and gatekeeper (if present)- allows terminals to joinleave zone requestreturnbandwidth provide status updates hellip

1048708 H245 (Call Control)- Media Control Protocol- Allows terminals to negotiate connection parameters (codec bit rate )

1048708 Q931 (Call Signalling)- Manages call setup and termination

SIP ndash Session Initiation Protocol1048708 Developed by IETF since 19991048708 RFC 2543 March 1999 (obsolete)1048708 RFC 3261 June 20021048708 Target develop simpler and more modular protocol for VoIP than the large and complex H323 by ITU1048708SIP is a text-based protocol similar to HTTP and SMTP for initiating interactive communication sessions between users1048708 SIP is an application-layer control (signalling) protocol for creating modifying and terminating sessions with one or more participants1048708 Sessions include Internet Multimedia conferences Internet Telephone calls and Multimedia distribution

SIP (2)1048708 SIP can be used with different transport protocols it doesnt even require reliable transport protocols1048708 A simple SIP client can be implemented using only UDP

SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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SIP ndash Session Initiation Protocol1048708 Developed by IETF since 19991048708 RFC 2543 March 1999 (obsolete)1048708 RFC 3261 June 20021048708 Target develop simpler and more modular protocol for VoIP than the large and complex H323 by ITU1048708SIP is a text-based protocol similar to HTTP and SMTP for initiating interactive communication sessions between users1048708 SIP is an application-layer control (signalling) protocol for creating modifying and terminating sessions with one or more participants1048708 Sessions include Internet Multimedia conferences Internet Telephone calls and Multimedia distribution

SIP (2)1048708 SIP can be used with different transport protocols it doesnt even require reliable transport protocols1048708 A simple SIP client can be implemented using only UDP

SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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SIP (2)1048708 SIP can be used with different transport protocols it doesnt even require reliable transport protocols1048708 A simple SIP client can be implemented using only UDP

SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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SIP (3)

Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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Comparison of H323 and SIP

Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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Comparison of H323 and SIP (2)

CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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CODECsSelecting the appropriate speech CODEC is essential CODEC performance includes the baseline quality (that is without impairments) and the performance with impairments present such as background noise and lost or late packets The table below shows some CODECs that are used for voice traffic Bandwidth requirements are estimates

CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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CODEC Selection

It is important to select a CODEC that meets the bandwidth and voice quality requirements

G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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G711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G711 does not compress the audio

G729 AB is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G711 and delivers near-toll voice quality for example a G729 AB with a 30 ms sample size is an effective technique to reduce bandwidth limitations while delivering acceptable voice quality

G7231 is used when bandwidth not voice quality is the customerrsquos objective

ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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ImportantSpeech CODECs (compression algorithms) such as G729 AB and G7231 are designed to reduce the bandwidth required however when using these CODECs consider parameters such as end-to-end delay (latency) and distortion in voice quality Although the G726 CODEC has less processing delay than the G729 AB CODEC G729 AB is generally used for VoIP because it delivers better voice quality

Tip Remember when selecting a CODEC consider the customerrsquosvoice quality targets and bandwidth requirements For example theG7231 CODEC might be appropriate if bandwidth consumption notvoice quality is the primary consideration However if voice qualitynot bandwidth consumption is the primary consideration anotherCODEC such as G711 is appropriate If bandwidth consumption ANDvoice quality are important consider G729 AB

VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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VoIP amp QoS

1048708 Voice quality characteristics- Clarity fidelity clearness and intelligibility of signal- Delay effect on interactivity- Echo distracting and confusing

1048708 Latency- Components Encoding Packetisation Network delayReceiver buffering Decoding- ITU-TG114 recommends 150ms

One-way Delay Effect on perceived Qualitylt100 -150ms Delay not detectable 150 -200ms Acceptible quality slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay normal conversation impossible

Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffers-Sudden changes in jitter may cause loss

Figure Playback buffer

Bandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G729 (AB) 8G722 48-64G711 64

Packet loss- Should be less then 5

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