The basics of multimedia exchange. VoIP and MultiMedia /76
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1 The basics of multimedia exchange VoIP and MultiMedia /76
2 What is this class about Encoding and transporting multimedia Signalling H.323, MGCP IAX MGCP XMPP SIP Security (TLS/CONNECT/ZRTP) NAT and firewall traversal (IPv6 and telephony) Deployment VoIP and MultiMedia /76
3 Where did it all start? VoIP and MultiMedia /76
4 Where did it all start? VoIP and MultiMedia /76
5 Encoding Media - PCM Pulse Code Modulation - a digital representation of an analog signal where the magnitude of the signal is sampled regularly at uniform intervals, then quantized to a series of symbols in a digital (usually binary) code. PCM has been widely used in: digital telephone systems compact disc red book format Computer systems (wav files). DVD or DVR Many Blu-ray Disc and HD-DVD movies Audio transmission within LANs Not used in real-time communication over the Internet due to high bandwidth consumption. VoIP and MultiMedia 2012 emil.ivov@jitsi.org 5/76
6 Encoding Media - PCM Sampling and quantization of a signal (red) for 4-bit PCM VoIP and MultiMedia 2012 emil.ivov@jitsi.org 6/76
7 Voice Codecs Codec Bit-Rate G kbps G ,3 kbps G729A ilbc 8 kbps 15.2 kbps Speex variable VoIP and MultiMedia 2012 emil.ivov@jitsi.org 7/76
8 Transporting media Let s first check out the basics though VoIP and MultiMedia 2012 emil.ivov@jitsi.org 8/76
9 <FRENCH> <! courtesy of Nicolas Montavont ENST Bretagne--> VoIP and MultiMedia /76
10 Modèles ISO et Internet Architectures de couches protocolaires : (possible) correspondance approximative ISO (OSI) Internet (TCP/IP) 7 Application 6 Présentation Application 5 4 Session Transport Transport communication de bout en bout 3 Réseau Réseau acheminement (routage) entre nœuds du réseau 2 Liaison Liaison accès au réseau physique 1 Physique Physique VoIP and MultiMedia 2012 emil.ivov@jitsi.org 10/76
11 Exemples de protocoles protocoles de routage Application OSPF RIP http ping Transport UDP TCP Réseau ARP IP ICMP Liaison Physique Ethernet IEEE (+ LLC + SNAP) PPP VoIP and MultiMedia 2012 emil.ivov@jitsi.org 11/76
12 Réseaux IP: encapsulation des données Données application bloc de données Application en-tête TCP bloc de données TCP en-tête IP en-tête TCP bloc de données IP en-tête Ethernet en-tête IP en-tête TCP bloc de données CRC Ethernet Pilote Ethernet Physique VoIP and MultiMedia /76
13 Réseaux IP: démultiplexage des données Exemple : TCP/ IP dans un réseau local Ethernet RIP UDP 520 http TCP 80 En-tête TCP ou UDP : champs N de port source et destination (+ adresse IP source) ICMP En-tête IP : champ Protocole En-tête Ethernet : champ Type IP ARP Ethernet Problème : comment faire remonter les données au long de la pile de protocoles? Solution : les en-têtes contiennent information permettant «l aiguillage» VoIP and MultiMedia 2012 emil.ivov@jitsi.org 13/76
14 Réseaux IP: Interconnexion de réseaux hétérogènes équipement terminal serveur web TCP protocole http (niveau applicatif) protocole TCP (de bout en bout) équipement terminal client web TCP IP protocole IP routeur (nœud) IP protocole IP IP protocole Ethernet Ethernet Ether. AP protocole a/b/g/n PPP LAN Ethernet liaison sans-fil VoIP and MultiMedia /76
15 Internet et IP Internet = ensemble de réseaux (Autonomous Systems ou AS) connectés entre eux Internetworking = interconnexion de réseaux AS = domaine administrative IP = Internet protocol Deux versions incompatibles entre elles IPv4 : version la plus courante aujourd hui IPv6 : IP «nouvelle génération» Un service à datagrammes Service non fiable Perte de paquets : possible Duplication de paquets : possible Arrivée des paquets en séquence : non garantie On parle également de service best-effort VoIP and MultiMedia 2012 emil.ivov@jitsi.org 15/76
16 ietf.org VoIP and MultiMedia /76
17 Internet quelques principes architecturaux Extrait du RFC 1958 par [Tanenbaum 2002] : Assurez-vous que ça marche Testez avant de finaliser la norme Préférez toujours la solution la plus simple Appliquez le Rasoir d Ockham S il y a plus d une manière de faire une chose, en choisissez une Évitez une multiplicité d options et paramètres Exploitez la modularité Pile protocolaire avec indépendance entre couches Préparez-vous à la diversité Flexibilité pour faire face à des équipements très hétérogènes VoIP and MultiMedia 2012 emil.ivov@jitsi.org 17/76
18 Réseaux IP et l Internet : normalisation ISOC (Internet Society) IAB (Internet Architecture Board) IANA (Internet Assigned Numbers Authority) IESG (Internet Engineering Steering Group) IRSG (Internet Research Steering Group) IETF (Internet Engineering Task Force) IRTF (Internet Research Task Force) Area WG WG... Area WG WG Research group RG WG WG RG etc.... VoIP and MultiMedia 2012 emil.ivov@jitsi.org 18/76
19 Domaines (Areas) Ex. : Transport, Internet Groupes de travail (Working Groups) Ex. : avt, dispatch, alto Principe de base : «rough consensus and running code» Participation aux décisions de normalisation : ouverte à tout le monde 3 réunions par an Listes de diffusion Création de groupes de travail VoIP and MultiMedia 2012 emil.ivov@jitsi.org 19/76
20 Normalisation à l IETF Accès libre et gratuit à tous les documents : Documents Normes : Request For Comments Ex. : RFC 793 (TCP) Caractère permanent Documents de travail : Internet Draft Ex. : draft-ietf-tsvwg-tcp-eifel-alg-07.txt (algorithme Eifel pour TCP) Caractère éphémère (6 mois de validité) VoIP and MultiMedia 2012 emil.ivov@jitsi.org 20/76
21 Normalisation à l IETF Processus de normalisation Standards track : Soumission d un draft personnel : draft-untel-mon-sujet-favori-00.txt à : internet-drafts@ietf.org Discussion dans les listes de diffusion et les réunions IETF Faire adopter le draft comme working group item : draft-ietf-xxxwg-mon-sujet-favori-00.txt Atteindre un consensus sur la liste de diffusion (last call) Donner le document à un Area Director Last call dans tous les groupes Si acceptation : envoi au RFC Editor (et à l IANA, si besoin d allouer des valeurs protocolaires) RFC : proposed standard, puis draft standard et enfin standard VoIP and MultiMedia 2012 emil.ivov@jitsi.org 21/76
22 Les RFC Classes de RFC : Documents issus du processus de normalisation (proposed standard, draft standard, standard) Documents non issus du processus de normalisation Experimental BCP (Best Current Practice) FYI (For Your Information)... Attention à la date! RFC 1149 (1er avril 1990) : A Standard for the Transmission of IP Datagrams on Avian Carriers RFC 2549 (1er avril 1999) : IP over Avian Carriers with Quality of Service RFC 3251 (1er avril 2002) : Electricity over IP RFC 3514 (1er avril 2003) : bit «paquet méchant» dans l en-tête IP VoIP and MultiMedia 2012 emil.ivov@jitsi.org 22/76
23 Évolution des normes En général, un protocole un RFC Exemple : TCP RFC 793 (spécification d origine) RFC 1122 (Requirements for Internet Hosts) RFC 1323 (Extensions for High Performance) RFC 2018, 2883 (Selective Acknowledgment) RFC 2581 (Congestion Control) RFC 2988 (Retransmission Timer) etc. etc.... VoIP and MultiMedia 2012 emil.ivov@jitsi.org 23/76
24 Activités de recherche : l IRTF Vision à plus long terme Des groupes parfois fermés Appartenance laissée au choix du chair Listes de diffusion publiques Quelques exemples P2P End-to-end Anti-spam Internet Measurement Optimisation pour le mobilité VoIP and MultiMedia 2012 emil.ivov@jitsi.org 24/76
25 </FRENCH> VoIP and MultiMedia /76
26 Where did it all start? VoIP and MultiMedia /76
27 Transporting Media - TCP vs. UDP Using UDP No way to detect loss. Order of delivery does not necessarily reflect the order of sending Using TCP Loss detection Respects order Loss recovery inefficient for CoIP Retransmission of lost segments increases jitter Decreasing window size causes lower bandwidth VoIP and MultiMedia 2012 emil.ivov@jitsi.org 27/76
28 UDP + RTP So, if both s**k, what do we do? Design a new transport protocol (SCTP) Design an application protocol that would compensate deficiencies of the transport protocol. A widespread solution Using an application protocol (RTP) over UDP VoIP and MultiMedia 2012 emil.ivov@jitsi.org 28/76
29 IETF Standardization 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 VoIP and MultiMedia 2012 emil.ivov@jitsi.org 29/76
30 OK, so what is RTP? RTP: A Transport Protocol for Real-Time Applications 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 VoIP and MultiMedia 2012 emil.ivov@jitsi.org 30/76
31 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 VoIP and MultiMedia 2012 emil.ivov@jitsi.org 31/76
32 Features of RTP and RTCP RTP Transports audio and video streams Describes the type of data it transports Adds timestamps and sequence numbers Does not allocate resources Does not do QoS RTCP Controls flows transported by RTP Exchange of basic information On the participants Quantative details over transmitted data VoIP and MultiMedia /76
33 The example of an audio conference How does the recipient determine the audio encoding of the flow The transmitting application sends the audio flow in separate blocs A format identifier in the RTP header How do we rebuild the audio stream? IP does not guarantee order and latency Using timestamps and sequence numbers in RTP headers How does a transmitter adapt to the bandwidth of the recipient? Using RTCP reports sent from the recipient to the sender. VoIP and MultiMedia 2012 emil.ivov@jitsi.org 33/76
34 RTP / RTCP Architecture RTP and RTCP operate over the transport layer. Could work independently of the underlying technology IP (UDP, TCP, SCTP), ATM Telephony over IP RTP ( used to ) use pair UDP ports RTCP ( used to) use odd UDP ports (rtp.port + 1) VoIP and MultiMedia 2012 emil.ivov@jitsi.org 34/76
35 Audio Session Channels Alice Bob Port xxxxx Port Port xxxxx Port RTP Flow RTP Flow RTCP Flow RTCP Flow RTCP Flows transport sender and receiver reports Port Port xxxxx Port Port xxxxx VoIP and MultiMedia /76
36 Audio Video Session Channels Alice Bob Port xxxxx Port Port xxxxx Port Port xxxxx Port Port xxxxx Port RTP Flow (audio) RTP Flow (audio) RTP Flow (video) RTP Flow (video) RTCP Flow (audio) RTCP Flow (audio) RTCP Flow (video) RTCP Flow (video) Port Port xxxxx Port Port xxxxx Port Port xxxxx Port Port xxxxx VoIP and MultiMedia /76
37 RTP Header V=2 P X CC M PT sequence number timestamp synchronization source (SSRC) identifier +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ contributing source (CSRC) identifiers VoIP and MultiMedia 2012 emil.ivov@jitsi.org 37/76
38 RTP Header (continued) 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. VoIP and MultiMedia 2012 emil.ivov@jitsi.org 38/76 emil.ivov@jitsi.org 38
39 RTP Header (end) 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 identifying the source of the packet. CSRC Identifier (32 bits) An integer identifying an SSRC contributing to the aggregated flow. Optional (only when mixing). VoIP and MultiMedia 2012 emil.ivov@jitsi.org 39/76 emil.ivov@jitsi.org 39
40 Examples of some codecs Audio G.711 (64 kbit/s) G.729 (8 kbit/s) G.723 (6,3 and 5,3 kbit/s) G.728 (16 kbit/s) G.726 (32 kbit/s) your own codec in here Video H.261 (40 kbit/s to 2 Mbit/s = f(resolution, frames per second, color depth)) H.263 MPEG4 JPEG VoIP and MultiMedia 2012 emil.ivov@jitsi.org 40/76 emil.ivov@jitsi.org 40
41 Some payload types (RFC 3551) PT encoding media type clock rate channels name (Hz) 0 PCMU A 8, GSM A 8, G723 A 8, DVI4 A 8, DVI4 A 16, LPC A 8, PCMA A 8, G722 A 8, L16 A 44, L16 A 44, QCELP A 8, CN A 8, MPA A 90,000 (see text) 15 G728 A 8, DVI4 A 11, DVI4 A 22, G729 A 8,000 1 dyn G729D A 8,000 1 dyn G729E A 8,000 1 VoIP and MultiMedia 2012 emil.ivov@jitsi.org 41/76
42 Bandwidth required by some codecs Format Payload Bandwidth (header only) Packet length Payload size Bandwidth (without hdr) Bandwidth (hdr) G KB/s 20 ms 160 bytes 80 Kb/s 64.8 Kb/s G KB/s 10 ms 80 bytes 96 Kb/s 65.6 Kb/s G KB/s 20 ms 20 bytes 24 Kb/s 8.8 Kb/s G KB/s 10 ms 10 bytes 40 Kb/s 9.6 Kb/s VoIP and MultiMedia 2012 emil.ivov@jitsi.org 42/76 emil.ivov@jitsi.org 42
43 Mixers and Translators Mixer Receives multiple flows from different sources called SSRCs (Synchronization SouRCes) Modifies their encoding if necessary Sends an aggregated flow. The initial SSRCs become CSRCs (Contributing SouRCe) Reason? Conference Calls Translators Receives multiple flows from different sources called SSRCs (Synchronization SouRCes) Modifies their encoding format and bandwidth Useful for handling NAT and firewall traversal Resends the flows as they were. VoIP and MultiMedia /76
44 A bit more on mixers SSRC 1 SSRC 2 SSRC M Contains: CSRC 1 CSRC 2 CSRC 3 SSRC 3 VoIP and MultiMedia 2012 emil.ivov@jitsi.org 44/76
45 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, , phone #) BYE : Ends an RTP session Leaving a conf call APP : Application specific packet Signaling specific for a particular application VoIP and MultiMedia 2012 emil.ivov@jitsi.org 45/76
46 Measuring transmission quality Regularly exchange send and receive 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 using application The using application is then supposed to adapt to call conditions. VoIP and MultiMedia 2012 emil.ivov@jitsi.org 46/76
47 Send Report Message Format (SR) 0! ! 1! ! 2! ! V=2! RC! PT=SR=200! Length! 3! 0 1! SSRC of sender! NTP Timestamp (most significant word)! NTP Timestamp (least significant word)! RTP Timestamp! Sender s packet count! Sender s octet count! SSRC_1 (SSRC of first source)! Fraction Lost Cumulative number of packets lost! Extended highest sequence number received! Inter-arrival jitter! Last SR (LSR)! Delay since last SR (DLSR)! SSRC_2 (SSRC of second source)!! Profile specific extensions! VoIP and MultiMedia 2012 emil.ivov@jitsi.org 47/76
48 Receive Report Message Format (RR) 0! ! 1! ! 2! ! V=2! RC! PT=RR=201! Length! 3! 0 1! Fraction Lost! SSRC of packet sender! SSRC_1 (SSRC of first source)! Cumulative number of packets lost! Extended highest sequence number received! Inter-arrival jitter! Last SR (LSR)! Delay since last SR (DLSR)! SSRC_2 (SSRC of second source)!! Profile specific extensions! VoIP and MultiMedia 2012 emil.ivov@jitsi.org 48/76
49 SDES Message Format 0! ! 1! ! 2! ! V=2! RC! PT=SDES=202! Length! 3! 0 1! SSRC/CSRC_1! SDES Items!! SSRC/CSRC_2! SDES Items! VoIP and MultiMedia 2012 emil.ivov@jitsi.org 49/76
50 SDES Items 0! ! 1! ! 2! ! 3! 0 1! Name = 2! Length! Variable length user name! = 3! Length! Variable length address! Phone = 4! Length! Variable length phone number! Location = 5! Length! Variable length user location! App tool = 6! Length! Name of the application! Note = 7! Length! Form-free note about the source! VoIP and MultiMedia 2012 emil.ivov@jitsi.org 50/76
51 BYE Packets 0! ! 1! ! 2! ! V=2! RC! PT=SDES=203! Length! 3! 0 1! Length! SSRC/CSRC!! Reason for leaving (optional) VoIP and MultiMedia 2012 emil.ivov@jitsi.org 51/76
52 Something working IP Address IP Address Problem non user friendly Unreliable Doesn t work behind NATs VoIP and MultiMedia 2012 emil.ivov@jitsi.org 52/76
53 The basics of IP telephony. network core (registrars, proxies, ) Bob Address: B Port: Pb Alice Address: A Port: Pa VoIP and MultiMedia 2012 emil.ivov@jitsi.org 53/76
54 The basics of IP telephony. network core (registrars, proxies, ) Bob Address: B Port: Pb Alice Address: A Port: Pa VoIP and MultiMedia 2012 emil.ivov@jitsi.org 54/76
55 The basics of IP telephony. network core (registrars, proxies, ) MEDIA Bob Address: B Port: Pb Alice Address: A Port: Pa VoIP and MultiMedia 2012 emil.ivov@jitsi.org 55/76
56 Signaling Popular signaling protocols. H.323 MGCP SIP XMPP/Jingle Inter-Asterisk exchange (IAX) Other signaling protocols: Skype, ICQ, Yahoo VoIP and MultiMedia /76
57 MEGACO- H248-MGCP MGCP - Media Gateway Control Protocol Defines protocols for control of gateways that handle media flow conversion Example: transcoding analogous voice (PSTN) into digital signal IP. This approach is based on the notion of separating signaling from multimedia support. VoIP and MultiMedia 2012 emil.ivov@jitsi.org 57/76
58 H248-MGCP (MEGACO) Media Gateway Controller PSTN IP Media Gateway PCM Flow 64 Kb/s RTP Flow VoIP and MultiMedia /76
59 H.323 The ITU solution for video conferencing on data networks: IP, ATM, Strongly inspired by the RNIS H320 standards for conferencing. Multiple ITU PSTN low band protocols are employed by H.323 Q.931 Supplementary services coming from Q.932 VoIP and MultiMedia /76
60 H.323 ITU Recommendations Version : Video telephony system for LANs with no QoS Version 4 November 2000 Pros and Cons: Compatible with H320 (PSTN) High complexity, difficult to adapt to the Internet (Firewall, NATs, QoS) VoIP and MultiMedia 2012 emil.ivov@jitsi.org 60/76
61 H.323 Entities Terminal Terminal Gatekeeper IP Network MCU Gateway RTC RNIS VoIP and MultiMedia /76
62 Architectures and protocols RAS : Registration Admission Status Gatekeeper registration Q.931 : signaling call Allows opening an H.245 connection H.245 : control call Information exchange (codec, address, RTP and RTCP port numbers) Activates channels VoIP and MultiMedia 2012 emil.ivov@jitsi.org 62/76
63 Ports Port Type Used for 389 static TCP ILS Registration (LDAP) 1300 static TCP H.235 Secure Signaling 1503 static TCP T static UDP Gatekeeper Discovery 1719 static UDP Gatekeeper RAS 1720 static TCP Q.931 Call Setup dynamic TCP H245 Control Channel dynamic UDP RTP/RTCP Audio/Video Streams VoIP and MultiMedia /76
64 A basic 2-party call UDP Q.931 port TCP 1720 H.245 port TCP > 1024 RTP G7xx RTCP RTP H26x RTCP RTP G7xx RTCP RTP H26x RTCP VoIP and MultiMedia /76
65 The Zone T T T GW GW SCN MCU GK GW VoIP and MultiMedia 2012 emil.ivov@jitsi.org 65/76
66 A Single Administrative Domain BE VoIP and MultiMedia /76
67 Multiple Administrative Domains Clearing House Packet Network VoIP and MultiMedia /76
68 CoIP Classes A few words about IAX VoIP and MultiMedia 2012 emil.ivov@jitsi.org 68/76
69 CoIP Classes A few words about XMPP/Jingle VoIP and MultiMedia 2012 emil.ivov@jitsi.org 69/76
70 CoIP Classes The Session Initiation Protocol SIP VoIP and MultiMedia /76
71 The Session Initiation Protocol Some of the People Behind It Henning Schulzrinne Department of Computer Science Columbia University, New York, USA Jonathan Rosenberg Cisco Systems VoIP and MultiMedia /76
72 Short History Developments of SIP falls under MMUSIC within IETF Multiparty Multimedia Session Control (MMUSIC) February 1996 Session Invitation Protocol (SIPv1) Internet Draft Mark Handley & Eve Schooler Purpose was to invite registered users to conference sessions Specified SDP and UDP Simple Conferencing Invitation Protocol (SCIP) Internet Draft Henning Schulzrinne Purpose was to invite users to point to point and multicast sessions Used identifiers, TCP, but defined its own format for session description December 1996 Session Initiation Protocol (SIPv2) Internet Draft Handley, Scholler & Schulzrinne HTTP based, could use UDP or TCP, and SDP for session description Jonathan Rosenberg became co-author in 1998 February 1999 SIP became a proposed standard, published as RFC 2543 VoIP and MultiMedia 2012 emil.ivov@jitsi.org 72/76
73 Short History March 2001: SIP Working group split: SIP Fundamental specification and its extensions SIPPING Applications that use SIP Notification services added later: SIMPLE IETF WG for Instant Messaging and Presence using SIP June 2002 new version published: RFC3261 obsoletes RFC 2543 Many other RFCs and Internet Drafts introduce extensions to SIP or standardise usage and interworking, e.g.: RFC 3263 SIP: Locating SIP Servers RFC 3265 SIP-Specific Event Notification RFC 3361 DHCP Option for SIP Servers RFC 3428 Session Initiation Protocol Extension for Instant Messaging RFC 3515 The Session Initiation Protocol (SIP) Refer Method Previously active working groups: SIP, SIPPING, SIMPLE Currently active working groups: DISPATCH, SIPCORE, SIMPLE, P2PSIP, SIPCLF VoIP and MultiMedia /76
74 The Session Initiation Protocol The Session Initiation Protocol (SIP) is an application-layer control protocol that can establish, modify, and terminate different kinds of sessions such as Internet telephony calls Request/response protocol (like HTTP) Uses a <header:value> format (like SMTP) Simple and extensible Designed for mobility (proxy/redirect servers) Authentication Capability negotiation Works on any transport: UDP, TCP, SCTP, ATM SIP is used for signaling: Instant Messaging sessions Phone calls over the Internet Gaming servers VoIP and MultiMedia 2012 emil.ivov@jitsi.org 74/76
75 Intra Domain SIP Signaling 4 thomas@u-strasbg.fr 5 SIP Soft Client 3 7 emcho@u-strasbg.fr 6 2 u-strasbg.fr Proxy Server 1 SIP Phone u-strasbg.fr Registrar and Location Service 1. Call Thomas - INVITE 2. Query Where is Thomas@u-strasbg.fr? (non-sip) 3. Response (non-sip) 4. Proxied Call - INVITE 5. Response - OK 6. Response - OK 7. Multimedia Chanel Establised RTP Streams VoIP and MultiMedia 2012 emil.ivov@jitsi.org 75/76
76 The basics of multimedia exchange VoIP and MultiMedia /76
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