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Voice over IP Nicolas Montavont [email protected] June 2013 2013 page VoIP - RSM department References ! Books Olivier Hersent , David Gurle , Jean-Pierre Petit , La voix sur IP : Déploiement des architectures VoIP, IMS et TISPAN Protocoles SIP 3GPP et IETF, H.323, MGCP Alan B. Johnston, SIP Understanding the Session Initiation Protocol, second edition, ARtech House Publishers Vineet Kumar, Markku Korpi, Sentil Sengodan, IP Telephony with H.323, Architecture for Unified Networks and Integrated Services, Wiley Ken Camp, IP Telephony demystified, McGraw-Hill Networking ! Wesites https://wiki.rsm.enstb.fr/eleves/index.php/RES202 http://www.tech-invite.com http://www.packetizer.com / http://www.asteriskguru.com 2

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Voice over IP

Nicolas [email protected]

June 2013

2013page VoIP - RSM department

References

! Books• Olivier Hersent, David Gurle, Jean-Pierre Petit, La voix sur IP : Déploiement

des architectures VoIP, IMS et TISPAN Protocoles SIP 3GPP et IETF, H.323, MGCP

• Alan B. Johnston, SIP Understanding the Session Initiation Protocol, second edition, ARtech House Publishers

• Vineet Kumar, Markku Korpi, Sentil Sengodan, IP Telephony with H.323, Architecture for Unified Networks and Integrated Services, Wiley

• Ken Camp, IP Telephony demystified, McGraw-Hill Networking! Wesites• https://wiki.rsm.enstb.fr/eleves/index.php/RES202• http://www.tech-invite.com• http://www.packetizer.com/• http://www.asteriskguru.com

2

2013page VoIP - RSM department

Voice over IP - program

3

!Introduction • To telephony application• To telephony network

!Codec!Transport of the voice

2013page VoIP - RSM department

Services de télécommunication

!Un service de télécommunication est une application informatique permettant pour différents utilisateurs distribués géographiquement, un partage d’informations opéré selon différentes modalités :• Structure du partage : en un coup (« one-shot ») / session• Synchronisme du partage : asynchrone / synchrone• Objet du partage : monomédia / multimédia

synchronismestructure asynchrone synchrone

one-shotTexte : SMS, forum, blogging, Image : MMSVoix/vidéo : messagerie vocale/vidéoTexte/image/voix/vidéo : mail, web

Texte : IM, micro-bloggingVoix/vidéo : webradio, « streaming »

session Texte/image/voix/vidéo : google wave Texte : IM, Voix/vidéo : téléphone, conférence

4

2013page VoIP - RSM department

Service téléphonique

!L’initiateur du service établit une communication, envoie et reçoit des informations et termine explicitement la communication

!Caractéristiques• Distribution géographique des participants• Durée d’établissement de la communication « quasi »

instantanée• Interactivité très forte entre les participants permettant à chacun

d’exploiter la présence dans la communication des autres interlocuteurs, tout en masquant leur absence physique

! Importance• De la joignabilité (un utilisateur peut être appelé à tout moment,

service push par essence)• De la mobilité

!Fait partie de la catégorie Service conversationnel

5

2013page VoIP - RSM department

Caractéristique du service téléphonique

!Une exigence forte !Un délai de transit de bout en bout, des données < 150 ms

pour garantir une interactivité de type conversationnelle

!Classes d’interactivité selon la recommandation G.114 de l’ITU-T [G.114]

Délai par sens Conséquence

0 à 150 ms Acceptable

150 à 300 ms Acceptable en cas de faible interactivité

300 à 700 ms Du type « talkie-walkie »

Au-delà de 700 ms Très difficile sans entrainement spécifique

6

2013page VoIP - RSM department

Structure du service téléphonique

!Établissement de la communication (ou de la session en téléphonie sur IP)

!Échanges de flux d’informations• Voix pour le téléphone

!Libération de la communication (ou de la session)

Alice Bob

Réseauétablissement

libération

échanges

7

2013page VoIP - RSM department

Now, what is Voice over IP?

!Transport voice over an IP network! Interactive application• Quality of service issues (a service over a non dedicated

architecture)!Two separate planes• Data plane• Signaling plane

!Strong need for a session protocol• Users establish sessions before exchanging data

!User location and mobility support

8

2013page VoIP - RSM department

What is an application layer protocol?

9

Application

Transport

Network

MAC

Physical

Application

Presentation

Session

OSI model

TCP/IP stack

2013page VoIP - RSM department

Application layer protocols

!Network services• Domain Name System

- Name resolution into IP address

!Network utilities• ARP - Address Resolution Protocol• FTP - File Transfer Protocol• SNMP - Simple Network Management Protocol

!Well-known Applications• HTTP• SMTP / IMAP / POP

10

2013page VoIP - RSM department

The big picture

!Transporting digitalized audio and video: RTP!Optimizing traffic (QoS): IntServ, DiffServ, MPLS

!Transporting signaling: UDP, TCP, SCTP!Signaling: • SIP• H.323• MGCP• IAX• XMPP - Jingle

11

2013page VoIP - RSM department

Goals for the ToIP

!Better usage of resources• Mutualisation of network resources (a unique backbone, a unique

WAN)• Low rate coding (lower than PCM)

!Decrease the cost• Mutualisation of equipments (« IPBX »= router + PABX) • Different billing for telephony and data (amount / duration, local /

distant)!Services convergence• Mobile services (voice and data)• Advanced telephony services

12

2013page VoIP - RSM department

Historic of ToIP/VoIP! Pre-commercial phase (before 1995)• 2 IETF working groups: AVT (Audio/Video Transport) et MMUSIC (Multiparty MUltimedia SessIon

Control) which specified RTP et SIP• Audio and video applications over the Internet (freeware, shareware)

! Commercial Phases « telephony over IP » (between 1995 and 1998)• Vocaltec offer in 1995• Call from a desktop to another desktop• Proprietary approaches (signaling and control)

! « Interworking IP / PSTN » phase (since 1998)• First version of H.323 since 1998, SIP came later (first RFC in 2000), H.248 came even later• Conformance with the ISDN signaling standards addressed in TIPHON• Toward a generalization of services• Desktop to phone, phone to desktop, phone to phone

! First home commercial offers form 2003• In parallel with the deployment of large band access at home• New actors (manufacturers, developers, integration, operators….)• Concurrence with non-normative approaches (SKYPE, ASTERISK…)

! VoIP as an element of the « Triple Play » offer• TV - Telephony - Internet

13

2013page VoIP - RSM department

ToIP challenges!QoS in term of transport• Delay (end-to-end delay, time for a packet to cross the network)• Jitter (measure of the variability over time of the packet latency across

a network. A network with constant latency has no variation (or jitter. Packet jitter is expressed as an average of the deviation from the network mean latency.)

• Loss (percentage of packets sent and never received)! Interoperability• Between the various architecture• Between the various manufacturers• Between the IP and PSTN networks (addressing, signaling)

!Scalability• No user at the beginning• Exponential growth to several billions of users in few users• Several domains of applications

14

Voice coding

2013page VoIP - RSM department

Voice coding

!The voice is in the [0;7000Hz] passband• Voice coding is usually between [300; 3400Hz]

!Sampling• Convert an analogic signal into a PAM signal• Transmission of instantaneous values of the signal• Nyquist frequency: Fs = 2 fmax

!Quantification• Convert the continuous values into discrete values• Logarithm scale: Compression law A or µ• The ratio signal over noise (C/I) gives the level of quality

!Coding

16

2013page VoIP - RSM department

Echantillonnage, Quantification, Codage

17

2013page VoIP - RSM department

Voice coding

18

• Waveform coding: Representation of the input signal without any a-priori knowledge of how the original signal was generated•Pulse Code Modulation (PCM)•Differential Pulse Code Modulation (DPCM)•Adaptative Differential Pulse Code Modulation (ADPCM)

• Vocoding: Usage of an explicit model of voice signal•Channel vocoder•Homomorphic vocoder

• Hybrid coding: combinaison of the two methods•Multi-Pulse Coding (MPC)•Sinusoidal Modeling (STC)

2013page VoIP - RSM department

PCM and beyond

����� ���

�"'362/4-�'9��1�>��/4*+6+4*+49�8'362+8����5-'7/9.3�8)'2+�54���(/98����!�,25<�'9� �1(/9�8

125µs

�/,,+7+49/'2��5*/4-

��"'362/4-�'9��1�>����5*/4-�9.+�*/,,+7+4)+8�(+9<++4�8:))+88/;+�8'362+8���5-'7/9.3�)5*/4-�</9.�2+88�9.'4���(/98����!�,25<��� �1(/9�8

7+*/)9/;+�)5*/4-

���5*/4-�5,�'�8+9�5,�;'2:+8�*+6+4*/4-�54�,:4)9/548����!���%�!�,25<���� �1(/9�8

19

2013page VoIP - RSM department

PCM G711 codec

!Sampling at 8kHz in the [300; 3400] passband!Linear quantification on 12 bits!Compression from 12 to 8 bits with the usage of a

logarithm scale with the A or µ law• Signal with low power are given more importance

!We get a digital signal rated at 64 kb/s, which usually produce a packet rate of 20ms• Each packet contains 80 samples

20

2013page VoIP - RSM department

Quality of a codec

!The reference is called toll quality, this is the quality of the G711 codec

!Important parameters to consider• Absolute quality of the signal

- Commercial parameter- Does not take into account the interactivity

• Delay introduced by the coding algorithm- Algorithmic delay- Linked to the size of the voice frame needed for the coding

• Complexity of the coding process• Behavior of the coding

- How does it code the music, the DTMF• Propriété de mise en cascade

- What is the impact of several encoding / decoding?• How does it react to errors?

21

2013page VoIP - RSM department

Quality of a codec…..

You will have to be very quiet.

There was nothing to be seen.

They worshipped wooden idols.

I want a minute with the inspector.

Did he need any money?

……..

……..

MOS Quality Impact

5 Excellent imperceptible

4 Good Perceptible Non gênante

3 Correct Peu gênante

2 bad gênante

1 very bad Très gênante22

Mean Opinion Score

2013page VoIP - RSM department

Source schéma : http://www-mobile.ecs.soton.ac.uk/

speech_codecs/common_classes.html

Qualité des types de codecs

23

2013page VoIP - RSM department

Usual voice codec

24

Codec Algorithm Rate Sample size (ms)

Delay of the codec (ms) MOS

G711 PCM 64 0.125 1 4.2

G726 ADPCM 16/24/32/40 0.125 1 2/3.2/4/4.2

G722 ADPCM 64 0.125 1 4.3

G728 CELP 16 0.625 5 4

G729 CS-ACELP 8 10 35 4

G723.1 ACELP-MP-MLQ 5.3 / 6.3 30 97.5 3.7 / 3.9

GSM RPE-LTP 13 20 3.8

iLBC LPC 15.2 / 13.3 20/30 70/85 4

Speex CELP 2.15-24.6 20 70

PSTN - Public Switched Telephony Network

2013page VoIP - RSM department

Le poste analogique

26

2013page VoIP - RSM department

Signalisations sur l’accès pour la numérotation téléphonique

!Anciennement : coupure de la ligne pendant quelques dizaines de ms • Chiffre 7 = 7 coupures de lignes successives

!Actuellement : Numérotation Q.23 ou DTMF!(Dual Tone Multifrequency) ou fréquence vocale!Appui touche = émission• fréquence colonne • + fréquence ligne

27

2013page VoIP - RSM department

Coeur de réseau: transmission par circuit ou paquet ?

Source slide: Kurose-Ross modifié

! Considérons un réseau de routeurs (au sens large)

! Question fondamentale : comment transmettre les données sur ce réseau?• Commutation de circuit : circuit

dédié par appel. Ex: réseau téléphonique dans sa version de base

• Commutation de paquet: des blocs (paquets) de données sont transmis de temps en temps (sporadique)

28

2013page VoIP - RSM department

PSTN architecture

29

2013page VoIP - RSM department

Principe du multiplexage temporel

!Time Division Multiple Access = TDMA• Multiplexage temporel

!En Europe, 32 intervalles de temps dont 2 réservés!Valeur de T ?

30

VoIP - Voice Transport

2013page VoIP - RSM department

The big picture (reminder)

!Transporting digitalized audio and video: RTP!Optimizing traffic (QoS): IntServ, DiffServ, MPLS

!Transporting signaling: UDP, TCP, SCTP!Signaling: • SIP• H.323• MGCP• IAX• XMPP - Jingle

32

2013page VoIP - RSM department

TCP vs. UDP

• UDP

33

2013page VoIP - RSM department

• UDP• No way to detect loss

34

TCP vs. UDP

2013page VoIP - RSM department

• UDP• No way to detect loss • Order of delivery does not necessarily reflect the order of sending

35

TCP vs. UDP

2013page VoIP - RSM department

• UDP• No way to detect loss • Order of delivery does not necessarily reflect the order of sending • No flow synchronization

36

TCP vs. UDP

2013page VoIP - RSM department

• UDP• No way to detect loss• Order of delivery does not necessarily reflect the order of sending • No flow synchronization

=> UDP can not reconstitute the audio flow !

37

TCP vs. UDP

2013page VoIP - RSM department

• UDP• No way to detect loss• Order of delivery does not necessarily reflect the order of sending • No flow synchronization

=> UDP can not reconstitute the audio flow !• TCP

38

TCP vs. UDP

2013page VoIP - RSM department

• UDP• No way to detect loss• Order of delivery does not necessarily reflect the order of sending • No flow synchronization

=> UDP can not reconstitute the audio flow !• TCP

• Loss detection

39

TCP vs. UDP

2013page VoIP - RSM department

• UDP• No way to detect loss• Order of delivery does not necessarily reflect the order of sending • No flow synchronization

=> UDP can not reconstitute the audio flow !• TCP

• Loss detection• Loss recovery

40

TCP vs. UDP

2013page VoIP - RSM department

• UDP• No way to detect loss• Order of delivery does not necessarily reflect the order of sending • No flow synchronization

=> UDP can not reconstitute the audio flow !• TCP

• Loss detection• Loss recovery

41

Retransmission of lost segmentsincreases jitter

Decreasing window sizecauses lower bandwidth

TCP vs. UDP

2013page VoIP - RSM department

• UDP• No way to detect loss• Order of delivery does not necessarily reflect the order of sending • No flow synchronization

=> UDP can not reconstitute the audio flow !• TCP

• Loss detection• Loss recovery•Respects order

42

TCP vs. UDP

Retransmission of lost segmentsincreases jitter

Decreasing window sizecauses lower bandwidth

2013page VoIP - RSM department

• UDP• No way to detect loss• Order of delivery does not necessarily reflect the order of sending • No flow synchronization

=> UDP can not reconstitute the audio flow !• TCP

• Loss detection• Loss recovery• Respects order•Packets transmission depends on the congestion window

43

TCP vs. UDP

Retransmission of lost segmentsincreases jitter

Decreasing window sizecauses lower bandwidth

2013page VoIP - RSM department

• UDP• No way to detect loss• Order of delivery does not necessarily reflect the order of sending • No flow synchronization

=> UDP can not reconstitute the audio flow !• TCP

• Loss detection• Loss recovery• Respects order• Packets transmission depends on the congestion window• No flow synchronization

44

TCP vs. UDP

Retransmission of lost segmentsincreases jitter

Decreasing window sizecauses lower bandwidth

2013page VoIP - RSM department

• UDP• No way to detect loss• Order of delivery does not necessarily reflect the order of sending • No flow synchronization

=> UDP can not reconstitute the audio flow !• TCP

• Loss detection• Loss recovery• Respects order• Packets transmission depends on the congestion window• No flow synchronization

=> TCP can not reconstitute the audio flow !

45

TCP vs. UDP

Retransmission of lost segmentsincreases jitter

Decreasing window sizecauses lower bandwidth

2013page VoIP - RSM department

Which transport protocol then?

46

!If both do not fit, what do we do?• Design a new transport protocol?• Design an application protocol that would compensate

deficiencies of the transport protocols?

2013page VoIP - RSM department

UDP + RTP

!If both do not fit, what do we do?• Design a new transport protocol?• Design an application protocol that would compensate

deficiencies of the transport protocols?!A widespread solution• Using an application protocol (RTP) over UDP

47

2013page VoIP - RSM department

! RFC 1889 (proposed standard), January 1996!H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson!RTP: A Transport Protocol for Real-Time Applications

! RFC 3550 (proposed standard), July 2003 (obsoletes 1889)!H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson!RTP: A Transport Protocol for Real-time Applications

10

IETF standardization

2013page VoIP - RSM department

OK, so what is RTP?

! The name is often used when referring to 2 related protocols!RTP = Real-time Transport Protocol!RTCP = Real-time Control Protocol

! Role!Provide a way of transporting data in a constant manner

under various time constraints.!Ex: audio and video flows

11

2013page VoIP - RSM department

Features of RTP and RTCP! RTP

!Transports audio and video streams!Describes the type of data it transports!Adds time stamps and sequence numbers!Does not allocate resources!Does not do QoS

! RTCP!Controls flows transported by RTP!Exchange of basic information

o On the participants o Quantitive details over transmitted data

12

2013page VoIP - RSM department

Channels for an audio session

Bob

RTP Flow Port 2222

Port xxxxx

RTP Flow Port xxxxx

Port 4444

RTCP Flow Port 2223

Port xxxxx

RTCP Flow Port xxxxx

Port 4445

RTCP Flows transport sender and receiver reports

51

Alice

2013page VoIP - RSM department

RTP header

52

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

|V=2|P|X| CC |M| PT | sequence number |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| timestamp |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| synchronization source (SSRC) identifier |

+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+

| contributing source (CSRC) identifiers |

| .... |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

2013page VoIP - RSM department

RTP Header

• Version (2 bits)• Current RTP version = 2

• Padding (1 bit)• If the padding bit is set, the packet contains one or more additional padding

octets at the end which are not part of the payload.• Extension (1 bit)

• If the extension bit is set, the fixed header MUST be followed by exactly one header extension.

• CSRC Count (4 bits)• The CSRC count contains the number of CSRC identifiers that follow the

fixed header.• Marker (1 bit)

• Depends on what we are carrying. It is intended to allow significant events such as frame boundaries to be marked in the packet stream.

53

2013page VoIP - RSM department

RTP Header

• Payload type (7 bits)• Identifies the format of the RTP payload and determines its interpretation

by the application• PT=0 for audio G.711 u-law 64 Kbit/s• PT=31 for video H.261

• Sequence Number (16 bits)• Arbitrary initial value, increased by 1 for every packet

• Timestamp (32 bits)• Time marker. Assists in determining delay and jitter.

• SSRC Identifier (32 bits)• An integer chosen randomly identifying the source of the packet.

• CSRC Identifier (32 bits)• An integer identifying an SSRC contributing to the aggregated flow.• Optional (only when mixing).

54

2013page VoIP - RSM department

Types de payload (RFC 3551)

6955

PT encoding media type clock rate channels

name (Hz)

___________________________________________________

0 PCMU A 8,000 1

3 GSM A 8,000 1

4 G723 A 8,000 1

5 DVI4 A 8,000 1

6 DVI4 A 16,000 1

7 LPC A 8,000 1

8 PCMA A 8,000 1

9 G722 A 8,000 1

10 L16 A 44,100 2

11 L16 A 44,100 1

12 QCELP A 8,000 1

13 CN A 8,000 1

14 MPA A 90,000 (see text)

15 G728 A 8,000 1

16 DVI4 A 11,025 1

17 DVI4 A 22,050 1

18 G729 A 8,000 1

dyn G729D A 8,000 1

dyn G729E A 8,000 1

2013page VoIP - RSM department

RTCP messages

!SR : Sender Report• Transmission statistics (bandwidth, loss, jitter, latency)

!RR : Receiver Report• Reception statistics (loss, jitter, latency)

!SDES : Source DEScription• Description of the transmitting party (name, email, phone #)

!BYE : Clears down an RTP session• Leaving a conf call

!APP : Application specific packet• Signaling specific for a particular application

56

2013page VoIP - RSM department

Measuring transmission quality

!Regularly exchange sender and receiver reports (SR & RR)• Every 5 seconds for low numbers of participants• Up to 5% of the total traffic for calls with a high number of

participants!Evaluation• Level of loss (%)• End to end latency • Jitter

!Objective• Supply information to the application • The application is then supposed to adapt to call conditions.

57

2013page VoIP - RSM department

Sender Report (SR)00 1 2 3 4 5 6 7 8 900 1 2 3 4 5 6 7 8 900 1 2 3 4 5 6 7 8 9

10 1 2 3 4 5 6 7 8 910 1 2 3 4 5 6 7 8 9

20 1 2 3 4 5 6 7 8 9

30 1

V=2 RC PT=SR=200PT=SR=200 LengthLengthLength

SSRC of senderSSRC of senderSSRC of senderSSRC of senderSSRC of senderSSRC of senderSSRC of sender

NTP Timestamp (most significant word)NTP Timestamp (most significant word)NTP Timestamp (most significant word)NTP Timestamp (most significant word)NTP Timestamp (most significant word)NTP Timestamp (most significant word)NTP Timestamp (most significant word)

NTP Timestamp (least significant word)NTP Timestamp (least significant word)NTP Timestamp (least significant word)NTP Timestamp (least significant word)NTP Timestamp (least significant word)NTP Timestamp (least significant word)NTP Timestamp (least significant word)

RTP TimestampRTP TimestampRTP TimestampRTP TimestampRTP TimestampRTP TimestampRTP Timestamp

Sender’s packet countSender’s packet countSender’s packet countSender’s packet countSender’s packet countSender’s packet countSender’s packet count

Sender’s octet countSender’s octet countSender’s octet countSender’s octet countSender’s octet countSender’s octet countSender’s octet count

SSRC_1 (SSRC of first source)SSRC_1 (SSRC of first source)SSRC_1 (SSRC of first source)SSRC_1 (SSRC of first source)SSRC_1 (SSRC of first source)SSRC_1 (SSRC of first source)SSRC_1 (SSRC of first source)

Fraction Lost Cumulative number of packets lostFraction Lost Cumulative number of packets lostFraction Lost Cumulative number of packets lostFraction Lost Cumulative number of packets lostFraction Lost Cumulative number of packets lostFraction Lost Cumulative number of packets lostFraction Lost Cumulative number of packets lost

Extended highest sequence number receivedExtended highest sequence number receivedExtended highest sequence number receivedExtended highest sequence number receivedExtended highest sequence number receivedExtended highest sequence number receivedExtended highest sequence number received

Inter-arrival jitterInter-arrival jitterInter-arrival jitterInter-arrival jitterInter-arrival jitterInter-arrival jitterInter-arrival jitter

Last SR (LSR)Last SR (LSR)Last SR (LSR)Last SR (LSR)Last SR (LSR)Last SR (LSR)Last SR (LSR)

Delay since last SR (DLSR)Delay since last SR (DLSR)Delay since last SR (DLSR)Delay since last SR (DLSR)Delay since last SR (DLSR)Delay since last SR (DLSR)Delay since last SR (DLSR)

SSRC_2 (SSRC of second source)SSRC_2 (SSRC of second source)SSRC_2 (SSRC of second source)SSRC_2 (SSRC of second source)SSRC_2 (SSRC of second source)SSRC_2 (SSRC of second source)SSRC_2 (SSRC of second source)

…………………

Profile specific extensionsProfile specific extensionsProfile specific extensionsProfile specific extensionsProfile specific extensionsProfile specific extensionsProfile specific extensions58

2013page VoIP - RSM department

Receive Report (RR)

00 1 2 3 4 5 6 7 8 900 1 2 3 4 5 6 7 8 900 1 2 3 4 5 6 7 8 900 1 2 3 4 5 6 7 8 9

10 1 2 3 4 5 6 7 8 910 1 2 3 4 5 6 7 8 9

20 1 2 3 4 5 6 7 8 9

30 1

V=2 RCRC PT=RR=201PT=RR=201 LengthLengthLength

SSRC of packet senderSSRC of packet senderSSRC of packet senderSSRC of packet senderSSRC of packet senderSSRC of packet senderSSRC of packet senderSSRC of packet sender

SSRC_1 (SSRC of first source)SSRC_1 (SSRC of first source)SSRC_1 (SSRC of first source)SSRC_1 (SSRC of first source)SSRC_1 (SSRC of first source)SSRC_1 (SSRC of first source)SSRC_1 (SSRC of first source)SSRC_1 (SSRC of first source)

Fraction LostFraction Lost Cumulative number of packets lostCumulative number of packets lostCumulative number of packets lostCumulative number of packets lostCumulative number of packets lostCumulative number of packets lost

Extended highest sequence number receivedExtended highest sequence number receivedExtended highest sequence number receivedExtended highest sequence number receivedExtended highest sequence number receivedExtended highest sequence number receivedExtended highest sequence number receivedExtended highest sequence number received

Inter-arrival jitterInter-arrival jitterInter-arrival jitterInter-arrival jitterInter-arrival jitterInter-arrival jitterInter-arrival jitterInter-arrival jitter

Last SR (LSR)Last SR (LSR)Last SR (LSR)Last SR (LSR)Last SR (LSR)Last SR (LSR)Last SR (LSR)Last SR (LSR)

Delay since last SR (DLSR)Delay since last SR (DLSR)Delay since last SR (DLSR)Delay since last SR (DLSR)Delay since last SR (DLSR)Delay since last SR (DLSR)Delay since last SR (DLSR)Delay since last SR (DLSR)

SSRC_2 (SSRC of second source)SSRC_2 (SSRC of second source)SSRC_2 (SSRC of second source)SSRC_2 (SSRC of second source)SSRC_2 (SSRC of second source)SSRC_2 (SSRC of second source)SSRC_2 (SSRC of second source)SSRC_2 (SSRC of second source)

……………………

Profile specific extensionsProfile specific extensionsProfile specific extensionsProfile specific extensionsProfile specific extensionsProfile specific extensionsProfile specific extensionsProfile specific extensions

59

2013page VoIP - RSM department

SN TS # données

RTP/UDP/IP

Transport with RTP

#���*+49/,=�9.+�)5*+)"����+26�/4�*+9+)9/4-�2588��7+�57*+7/4-�6')1+98#"���225<�95�7+)5489/9:9+�'�6+7/5*/)�,25<��.+26�/4�*+9+73/4/4-�*+2'=�'4*�0/99+7

"/>+�5,�9.+�.+'*+7�/4�� ;������� ����$� ����!# �60

2013page VoIP - RSM department

Effective rate of VoIP

30

20

20

20

20

Duration of the coded voice (ms)

16/17

24

32

32/40/48/56

80

Effective IP rate (kbps)

5,3/6,3 G.723.1

8G.729

16G.728

16/24/32/40G.726

64G.711

Codec rate(kbit/s)

Codec ITU

$� !# �

'=25'*��3:29/62+�8'362+8�;+7.+'*��.+'*+78

61

2013page VoIP - RSM department

VoIP: delay

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Traitement Durée en ms

Au départ

Mise en paquet 30

codage 35

En transmission

Délai réseau (sans « tromboning ») 60/120

En réception

récupération gigue 10/60

décodage 0

délai total 145/235

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62

2013page VoIP - RSM department

The playout

flux de paquets émis à débit constant(périodique)

Don

nées

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nsm

ises

t

Flux reçu flux de paquetsremis à débit constant(périodique)

client playoutdelay

buff

ered

data

Giguede

transport

dispersion

agrégation

63

Signaling protocol

2013page VoIP - RSM department

Control in VoIP

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65