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TRANSCRIPT
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
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Voice over IP - program
3
!Introduction • To telephony application• To telephony network
!Codec!Transport of the voice
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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
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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
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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
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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
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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
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What is an application layer protocol?
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Application
Transport
Network
MAC
Physical
Application
Presentation
Session
OSI model
TCP/IP stack
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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
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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
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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
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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
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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
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Voice coding
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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
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Echantillonnage, Quantification, Codage
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Voice coding
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• 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)
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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-
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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
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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?
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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
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Source schéma : http://www-mobile.ecs.soton.ac.uk/
speech_codecs/common_classes.html
Qualité des types de codecs
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Usual voice codec
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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
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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
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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)
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PSTN architecture
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Principe du multiplexage temporel
!Time Division Multiple Access = TDMA• Multiplexage temporel
!En Europe, 32 intervalles de temps dont 2 réservés!Valeur de T ?
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VoIP - Voice Transport
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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
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TCP vs. UDP
• UDP
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• UDP• No way to detect loss
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TCP vs. UDP
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• UDP• No way to detect loss • Order of delivery does not necessarily reflect the order of sending
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TCP vs. UDP
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• UDP• No way to detect loss • Order of delivery does not necessarily reflect the order of sending • No flow synchronization
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TCP vs. UDP
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• 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 !
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TCP vs. UDP
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• 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
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TCP vs. UDP
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• 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
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TCP vs. UDP
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• 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
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TCP vs. UDP
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• 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
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Retransmission of lost segmentsincreases jitter
Decreasing window sizecauses lower bandwidth
TCP vs. UDP
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• 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
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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
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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
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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 !
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TCP vs. UDP
Retransmission of lost segmentsincreases jitter
Decreasing window sizecauses lower bandwidth
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Which transport protocol then?
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!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?
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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
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! 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
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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
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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
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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
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Alice
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RTP header
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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 |
| .... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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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.
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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).
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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
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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
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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.
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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
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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
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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
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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
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VoIP: delay
#.+�959'2�*+2'=�(+9<++4�9.+�8+4*/4-�'4*�9.+�7+)+69/54�/8�
���+2'=8�*:+�95�9.+�)5*+)8�� ')1+9/>'9/54�5,�9.+�8'362+8��#7'486579�/4�9.+�45*+8��#7'486579�54�9.+�2/418��!+35;+�9.+�0/99+7
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
�:*-+9�,57�9.+�������)5*+)
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The playout
flux de paquets émis à débit constant(périodique)
Don
nées
tra
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