Protocols supporting VoIP

Transcription

1 Protocols supporting VoIP Dr. Danny Tsang Department of Electronic & Computer Engineering Hong Kong University of Science and Technology 1

2 Outline Overview Session Control and Signaling Protocol H.323 SIP Gateway Control and Signaling Protocol Media Gateway Control Protocol (MGCP) (RFC 3435) Megaco/H.248 Media Transport Protocol RTP/RTCP (RFC 1889) RTP payload for DTMF digits (RFC 2833) VoIP protocols 2

3 A snapshot of current VoIP protocols Standardized Proprietary IETF ITU ITU/IETF Open Source PBX Skinny more and more VoIP protocols 3

4 VoIP has to be standardized Proprietary protocols -> innovation is restricted Today s Internet offers diverse functionalities because of standardized communication protocols We can choose from SIP, H.323, or H.248/Megaco for VoIP signaling protocols H.248/Megaco: Used to control media gateways (between IP and PSTN) VoIP protocols 4

5 VoIP Standards Many different standards, e.g.: ITU-T H.323 etc. IETF SIP etc. ITU-T and IETF MGCP, Megaco/H.248 Vendor-specific standards But: over the last couple of years, SIP and the related IETF protocols got widespread acceptance in the market E.g. SIP, RTP, RTSP, RSVP etc. are supported by Windows XP out of the box VoIP protocols 5

6 Different approaches for VoIP protocol design VoIP is heavily standards-driven (interoperability) People working on standards for VoIP come from two different communities Traditional voice networks (bellheads) IP networking (netheads) Centralized vs. decentralized models of call control Bellheads tend to see terminals as stupid and networks as smart Netheads tend to see networks as stupid and terminals as smart Reflected to a certain extent in H.323 vs. SIP VoIP protocols 6

7 Protocols Needed for VoIP Signaling protocol (Call Control) To establish presence, locate users, set up, modify and tear down sessions Control protocol (Terminal Control) To negotiate codec capability, port number, etc. Media Transport Protocols Transmission of packetized audio/video Supporting Protocols QoS, Gateway Location, Interdomain Authentication, Authorization, Accounting, address translation, etc. VoIP protocols 7

8 Major VoIP Protocol Standards Session Control and Signaling H.323 H.225 Gateway Control and Signaling Media Transport Audio/ Video H.245 Q.931 RAS SIP MGCP Megaco/H.248 RTP RTCP RTSP TCP UDP IP H.323 Version 1 and 2 supports H.245 over TCP, Q.931 over TCP and RAS over UDP. H.323 Version 3 and 4 supports H.245 over UDP/TCP and Q.931 over UDP/TCP and RAS over UDP. SIP supports TCP and UDP. VoIP protocols 8

9 A quick comparison between H.323 and SIP SIP is a generic signaling protocol SIP only defines processes for address resolution and session establishment SIP is less complex than H.323 SIP is better suited for the integration of web-based applications with audio/video H.323 can only be used for A/V conferencing; T.120 additionally provides for data collaboration H.323 offers better interoperability (mandatory codecs, maturity) VoIP protocols 9

10 Outline Overview Session Control and Signaling Protocol H.323 SIP Gateway Control and Signaling Protocol Media Gateway Control Protocol (MGCP) (RFC 3435) Megaco/H.248 Media Transport Protocol RTP/RTCP (RFC 1889) RTP payload for DTMF digits (RFC 2833) VoIP protocols 10

11 Overview of Session Control and Signaling Protocol Session: something more general than a typical bidirectional telephone conversation E.g., multiparty video conference There exist many session control protocols; they share common functionalities VoIP protocols 11

12 Generic Session Control Caller Callee (1) Session setup request (2) I am working on it (3) Session setup confirmation (4) Negotiate terminal capabilities (5) Establish connection voice and/or video packets flow.. (6) End participation in session (7) Acknowledge end of session VoIP protocols 12

13 Simplistic H.323 session control flow Original terminal (1) H SETUP (2a) H CALL PROC (2b) H ALERTING (3) H CONNECT Destination terminal (4 and 5) H.245 Session establishment voice and/or video packets flow.. (6) H.245 Session release (7) H RLC VoIP protocols 13

14 H.323 uses H.225/RAS Signaling (Registration Admission Status) Terminal GateKeeper (1) H Registration Request (2) H Registration Confirmation (endpoint is registered) (3) H Admission Request (4) H Admission confirm (endpoint may place call) VoIP protocols 14

15 Simplistic SIP signaling flow Original terminal (1) INVITE (2) 180 Ringing (3) 200 OK Destination terminal ACK voice and/or video packets flow.. (6) BYE (7) 200 OK VoIP protocols 15

16 What is H.323? ITU-T Recommendation H.323 Version 5 (2003), current Version 7 (2009) Describes terminals and other entities that provide multimedia communications services over Packet Based Networks (PBN) which may not provide a guaranteed Quality of Service. H.323 entities may provide real-time audio, video and/or data communications. H.323 is an umbrella standard, not just a signaling protocol VoIP protocols 16

17 H.323 is an umbrella specification H.323: Infrastructure of audiovisual services Systems and terminal equipment for audiovisual services: Packet-based multimedia communications systems H.245: Control protocol for multimedia communication H.225: Call signalling protocols and media stream packetization for packet based multimedia communication systems Q.931: ISDN user-network interface layer 3 specification for basic call control H.235: Security and encryption for H-Series (H.323 and other H.245 based) multimedia terminals H.450.1: Generic functional protocol for the support of supplementary services in H.323 Codecs G.711: Pulse Code Modulation (PCM) of voice frequencies G.722: 7 khz audio-coding within 64 kbit/s G.723.1: Speech coders: Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s VoIP protocols 17

18 H.323 is an umbrella specification(2) More codecs: G.728: Coding of speech at 16 kbit/s using low-delay code excited linear prediction G.729: Coding of speech at 8 kbit/s using Conjugate Structure Algebraic-Code-Excited Linear-Prediction (CS-ACELP) H.261: Video codec for audiovisual services at p 64 kbit/s H.263: Video coding for low bit rate communication T.120: Data protocols for multimedia conferencing T.460: NAT/Firewall Traversal X.680: Information Technology - Abstract Syntax Notation One (ASN.1) - Specification of basic notation X.691: Information Technology - ASN.1 Encoding Rules - Specification of Packed Encoding Rules (PER) At least one audio channel is required - video is optional VoIP protocols 18

19 Entities The H.323 Architecture Terminals: LAN client endpoints Gateways: Interface between PSTN and the Internet Gatekeepers: control manager Multipoint Control Units (MCUs): provide conference capabilities H.323 Zone Collection of terminals, gateways, MCUs registered with a single gatekeeper. Protocols H consists of RAS and specifies Q.931 for call signaling H.245 for terminal capability negotiation (such as codec, etc) RTP/RTCP for media transport Audio/video codecs VoIP protocols 19

20 H.323 Components Processor Application for IP telephony Terminal (H.323) Terminal (H.320) PSTN ISDN H.323 Gateway Dedicated IP network Without guaranteed QoS (Internet/Intranet) H.323 Gateway PLMN PLMN: public land mobile network Gatekeeper (H.323) Multipoint Control Unit (H.323) VoIP protocols 20

21 H.323 Terminals Endpoint on a LAN Supports real-time, 2-way communications with another H.323 entity H.323 terminals are client endpoints that must support: H.225 call control signaling H.245 control channel signaling for terminal capability RTP/RTCP protocols for media packets Audio codecs Optional support: Video: H.261, H.263 Data: T.38, T.120 VoIP protocols 21

22 H.323 Terminals Video I/O Equipment Video Codec H.261, H.263 Audio I/O Equipment User Data Applications T.120, etc Audio Codec G.711, G.722 G.723, G.728 G.729 Receive Path Delay (jitter buffer) H Layer Local Area Network Interface System Control H.245 Terminal Control System Control User Interface Call Control H RAS Control H VoIP protocols 22

23 H.323 Gateway Interface between the LAN and the switched circuit network (SCN) A gateway provides translation: For example, a gateway can provide translation between entities in a packet switched network (eg. IP network) and Switched Circuit Network (example, PSTN network) Gateways can provide transmission formats and communication procedures translations, H.323 and Non - H.323 endpoints or codec translations VoIP protocols 23

24 H.323 Gatekeepers A gatekeepers is the most vital component of the H.323 system, dispatching the duties of a manager. Manage a zone (a collection of H.323 devices) Usually one gatekeeper per zone alternate gatekeeper might exist for backup and load balancing Typically a software application implemented on a PC, but can be integrated in a gateway or terminal Gatekeepers provide these functions: Address translation Admission control Call control signaling Call authorization Bandwidth management Call management VoIP protocols 24

25 H.323 Multipoint Control Unit Responsible for managing multipoint conferences (two or more endpoints engaged in a conference) Can be stand-alone device or integrated into a gateway, gatekeeper or terminal The MCU contains a mandatory Multipoint Controller (MC) that manages the call signaling optionally have Multipoint Processors (MPs) to handle media mixing, switching, or other media processing VoIP protocols 25

26 H.323 Protocol Stack Media H.323 Multimedia Applications, User Interface Video codecs - H.261, H.263, H.264 Audio codecs - G.711, G.723, G.729 Data Applications Media Control Terminal Control and Management Media over RTP/RTCP Audio Codecs Video Codecs Data/Fax T.120 Data conferencing T.38 Fax T.120 T.38 G.711 G G H.261 H.263 H RTCP H Q.931 Call Signaling H.245 H RAS RTP TCP TCP/UDP UDP TCP/UDP TCP UDP Call Control and Signaling H Capabilities advertisement, media channel establishment, and conference control. H.225 Q call signaling and call setup. RAS - registration and other admission control with a gatekeeper. IP VoIP protocols 26

27 ITU-T H.323 protocol summary Audio Codecs: G.711, G.722, G.723, G.729, GSM 6.10, etc. Call control, signaling: H.225, Q931 Path and parameter negotiation: H.245 Additions: Video Codecs: H.261, H.263, etc. Security, encryption, authentication: H.235 Supplementary services: H.450 (call transfer, call waiting, etc.) Gatekeeper and media gateway: RAS: registration, admission, status: H.225 Megaco/H.248: media gateway control protocol VoIP protocols 27

28 Information streams in H.323 Video Audio Data (T.120) Whiteboarding Pictures Any sort of shared data Communications control (H.245) Capabilities exchange Open/close logical channels Mode changes Call control (H.225) Call establishment Call tear-down VoIP protocols 28

29 Basic H.323 Call Gatekeeper A ACF LRQ LCF Gatekeeper B ACF RRQ/RCF IP Network RRQ/RCF ARQ V Gateway A H.225 (Q.931) Setup H.225 (Q.931) Alert and Connect H.245 RTP V ARQ Gateway B Phone A RRQ: Registration Request RCF: Registration Confirmation ARQ: Admissions Request ACF: Admissions Confirm LRQ: Location Request LCF: Location Confirmation Phone B VoIP protocols 29

30 H.323 protocol phases VoIP protocols 30

31 H.225 TCP connection on a well-known port Used to perform call signaling Also specifies packetization for all H.323 communication Call signaling is based on ISDN signaling (Q.931) Media are packetized using RTP (including RTCP control channel) Work on optional UDP connection on well-known port underway VoIP protocols 31

32 Registration, Admission, Status - RAS signaling Separate UDP-based H.225 stream Defined in H Functions: Allows an endpoint to request authorization to place or accept a call Allows a Gatekeeper to control access to and from devices under its control Allows a Gatekeeper to communicate the address of other endpoints Allows two Gatekeepers to easily exchange addressing information VoIP protocols 32

33 Registration, Admission, and Status RAS (cont) Terminal GateKeeper (1) RRQ: Registration Request (2) RCF: Registration Confirmation (endpoint is registered) (3) ARQ: Admissions Request (4) ACF: Admissions confirm (endpoint may place call) (5) DRQ: Disengage Request (call has terminated) (6) DCF: Disengage Confirmation VoIP protocols 33

34 H.245 Connection control function of H.323: Master/slave determination Capability Exchange Logical Channel Signaling Close Logical Channel Signaling Mode Request Round Trip Delay Determination Maintenance Loop Signaling May be used for transmitting user input, for example DTMF strings Encoded using ASN.1 PER VoIP protocols 34

35 ITU H.323 available source code: openh323.org, gnugk.org: gatekeeper, gateway, and terminal software (including G.711 audio) Ekiga.org (formerly Gnomemeeting.org): H.323 gatekeeper and terminal software (including LPC10, GSM, G.711, G.726 audio) VoIP protocols 35

36 Outline Overview Session Control and Signaling Protocol H.323 SIP Gateway Control and Signaling Protocol Media Gateway Control Protocol (MGCP) (RFC 3435) Megaco/H.248 Media Transport Protocol RTP/RTCP (RFC 1889) RTP payload for DTMF digits (RFC 2833) VoIP protocols 36

37 SIP Session Initiation Protocol Comes from IETF SIP long-term vision All telephone calls and video conference calls take place over the Internet People are identified by names or addresses, rather than by phone numbers. You can reach the callee, no matter where the callee roams, no matter what IP device the callee is currently using. VoIP protocols 37

38 SIP Architecture A signaling protocol The setup, modification, and tear-down of multimedia sessions SIP + SDP Describe the session characteristics Separate signaling and media transport VoIP protocols 38

39 SIP Network Entities Clients User agent clients Application programs sending SIP requests Servers Responds to clients requests Clients and servers may be in the same computer Proxy Acts as both clients and servers VoIP protocols 39

40 SIP Services Setting up a call Provides mechanisms for caller to let callee know she wants to establish a call Provides mechanisms so that caller and callee can agree on media type and encoding. Provides mechanisms to end call. Determine current IP address of callee. Maps mnemonic identifier to current IP address Call management Add new media streams during call Change encoding during call Invite others Transfer and hold calls VoIP protocols 40

41 Setting up a call to a known IP address Alice INVITE bob@ c=in IP m=audio RTP/AVP 0 port 5060 port OK c=in IP m=audio RTP/AVP 3 ACK port Bob Bob's terminal rings Alice s SIP invite message indicates her port number & IP address. Indicates encoding that Alice prefers to receive (PCM ulaw) Bob s 200 OK message indicates his port number, IP address & preferred encoding (GSM) m Law audio port GSM port SIP messages can be sent over TCP or UDP; here sent over RTP/UDP. time time Default SIP port number is VoIP protocols 41

42 Setting up a call (more) Codec negotiation: Suppose Bob doesn t have PCM ulaw encoder. Bob will instead reply with 606 Not Acceptable Reply and list encoders he can use. Alice can then send a new INVITE message, advertising an appropriate encoder. Rejecting the call Bob can reject with replies busy, gone, payment required, forbidden. Media can be sent over RTP or some other protocol. VoIP protocols 42

43 Example of SIP message INVITE SIP/2.0 Via: SIP/2.0/UDP From: To: Call-ID: Content-Type: application/sdp Content-Length: 885 c=in IP m=audio RTP/AVP 0 Notes: HTTP message syntax sdp = session description protocol Call-ID is unique for every call. Here we don t know Bob s IP address. Intermediate SIP servers will be necessary. Alice sends and receives SIP messages using the SIP default port number Alice specifies in Via: header that SIP client sends and receives SIP messages over UDP VoIP protocols 43

44 Name translation and user locataion Caller wants to call callee, but only has callee s name or address. Need to get IP address of callee s current host: user moves around DHCP protocol user has different IP devices (PC, PDA, car device) Result can be based on: time of day (work, home) caller (don t want boss to call you at home) status of callee (calls sent to voic when callee is already talking to someone) Service provided by SIP servers: SIP registrar server SIP proxy server VoIP protocols 44

45 SIP Registrar When Bob starts SIP client, client sends SIP REGISTER message to Bob s registrar server (similar function needed by Instant Messaging) Register Message: REGISTER sip:domain.com SIP/2.0 Via: SIP/2.0/UDP From: sip:bob@domain.com To: sip:bob@domain.com Expires: 3600 VoIP protocols 45

46 SIP Proxy Alice sends invite message to her proxy server contains address Proxy responsible for routing SIP messages to callee possibly through multiple proxies. Callee sends response back through the same set of proxies. Proxy returns SIP response message to Alice contains Bob s IP address Note: proxy is analogous to local DNS server VoIP protocols 46

47 Example Caller with places a call to keith@upenn.edu (1) Jim sends INVITE message to umass SIP proxy. (2) Proxy forwards request to upenn registrar server. (3) upenn server returns redirect response, indicating that it should try keith@eurecom.fr SIP proxy umass.edu 1 8 SIP client SIP registrar upenn.edu SIP registrar eurecom.fr 6 5 SIP client (4) umass proxy sends INVITE to eurecom registrar. (5) eurecom registrar forwards INVITE to , which is running keith s SIP client. (6-8) SIP response sent back (9) media sent directly between clients. Note: also a SIP ack message, which is not shown. VoIP protocols 47

48 Comparison with H.323 H.323 comes from the ITU (telephony). H.323 is another signaling protocol for real-time, interactive communication. H.323 is a complete, vertically integrated suite of protocols for multimedia conferencing: signaling, registration, admission control, transport and codecs. SIP comes from IETF: Borrows much of its concepts from HTTP. SIP has a Web flavor, whereas H.323 has a telephony flavor. SIP uses the KISS principle: Keep it simple stupid. SIP is a single component. Works with RTP, but does not mandate it. Can be combined with other protocols and services. VoIP protocols 48

49 IETF SIP available source code: linphone.org: SIP client (including LPC10, GSM, G.711) asterisk.org: SIP and H.323 terminal and gateway (including GSM, G.711, G.726 audio) VoIP protocols 49

50 Outline Overview Session Control and Signaling Protocol H.323 SIP Gateway Control and Signaling Protocol Media Gateway Control Protocol (MGCP) (RFC 3435) Megaco/H.248 Media Transport Protocol RTP/RTCP (RFC 1889) RTP payload for DTMF digits (RFC 2833) VoIP protocols 50

51 Signaling Gateway and Media Gateway Instead of PC-to-PC VoIP, there is also PC-to-Phone VoIP Gateway is needed to interconnect networks with different network protocol technologies by performing the required protocol conversions Traditional telephone systems Signaling (D - channel) -- used to control Media (B - channel) -- used to carry voice and data VoIP networks Signaling Gateway (SG) translate the D-channel data to VoIP-compatible signaling protocol, e.g. SIP or H.323 Media Gateway (MG) provide the bridge for media to seamlessly transit between PSTN and VoIP networks ferry media between B-channel and RTP steams Signaling path and Media path are different in VoIP systems. Signaling through H.323 gatekeepers (or SIP proxies) Media directly (end-to-end) VoIP protocols 51

52 Separation of Media and Call Control A network gateway has two related but separate functions. Signaling conversion The media gateway controller - MGC uses signaling to communicate with gateways Media conversion A slave function (mastered by call-control entities) Advantages of Separation Media conversion close to the traffic source and sink The call-handling functions are centralized. A call agent (media gateway controller - MGC) can control multiple gateways. New features can be added more quickly. VoIP protocols 52

53 Separation of Media and Call Control Signaling and Call Control control Signaling over IP Signaling and Call Control control Media Conversion Media over IP Media Conversion VoIP protocols 53

54 Gateway Control Protocol Born out of the need for IP networks to interwork with traditional telephony systems Provide control of media streams as they transmit between IP and traditional telephone networks Used by Media Gateway Controller (MGC) to control Media Gateways (MG) master-slave protocol Call agents (MGCs) control the operation of MGs Call-control intelligence Related call signaling MGs Do what the CA instructs A line or trunk on circuit-switched side to an RTP port on the IP side Communication between call agents Likely to be SIP VoIP protocols 54

55 MGC vs. MG MGC Contains all call control intelligence Forward Transfer Conference Hold Implements any peer-level protocols for interaction with other MGCs or peer entities Manage any interactions with signaling, such as SS7 MG Implements media connections to and from the packets based network (IP or ATM) Control gateway device features, such as user interface Has no knowledge of call level features, acts as a simple slave VoIP protocols 55

56 Media gateway control vs. call signaling SG MGC SIP-T, ISUP in H.323, Q.BICC MGC SG SIP SIP User Agent PSTN Gateway control protocol H.323 call signalling Gateway control protocol PSTN H.323 Endpoint MG MG Call signalling Media gateway control signalling Media flows VoIP protocols 56

57 Comparison between SIP, H.323 and MGCP SIP and H.323 Signaling Protocols Used to Set up and manage calls Relationship between entity Functional Entity Slave Entity peer to peer decentralized Master/slave protocol Peer Protocol Functional Entity Slave Entity Master/slave protocol MGCP Media Control Protocols Set up media streams and establish media paths between IP and other networks Relationship between Media Gateway Controller and Media Gateway master/slave centralized one MGC can be in charge of multiple MGs VoIP protocols 57

58 MGCP fundamentals Best-known media gateway control protocols MGCP Combination of SGCP and IPDC Standardized by IETF (RFC2705) Megaco/H.248 Extension of MGCP Megaco IETF (RFC3015) H.248 ITU Study Group 16 Other media gateway control protocols Skinny -- Cisco Simple Gateway Control Protocol (SGCP) Internet Protocol Device Control (IPDC) VoIP protocols 58

59 Outline Overview Session Control and Signaling Protocol H.323 SIP Gateway Control and Signaling Protocol Media Gateway Control Protocol (MGCP) (RFC 3435) Megaco/H.248 Media Transport Protocol RTP/RTCP (RFC 1889) RTP payload for DTMF digits (RFC 2833) VoIP protocols 59

60 Real-Time Transport Protocol (RTP) RTP specifies a packet structure for packets carrying audio and video data RFC RTP packet provides payload type identification packet sequence numbering timestamping RTP runs in the end systems. RTP packets are encapsulated in UDP segments Interoperability: If two Internet phone applications run RTP, then they may be able to work together VoIP protocols 60

61 RTP runs on top of UDP RTP libraries provide a transport-layer interface that extend UDP: port numbers, IP addresses payload type identification packet sequence numbering time-stamping VoIP protocols 61

62 RTP Example Consider sending 64 kbps PCM-encoded voice over RTP. Application collects the encoded data in chunks, e.g., every 20 msec = 160 bytes in a chunk. The audio chunk along with the RTP header form the RTP packet, which is encapsulated into a UDP segment. RTP header indicates type of audio encoding in each packet sender can change encoding during a conference. RTP header also contains sequence numbers and timestamps. VoIP protocols 62

63 RTP and QoS RTP does not provide any mechanism to ensure timely delivery of data or provide other quality of service guarantees. RTP encapsulation is only seen at the end systems: it is not seen by intermediate routers. Routers providing best-effort service do not make any special effort to ensure that RTP packets arrive at the destination in a timely matter. VoIP protocols 63

64 RTP Header Payload Type (7 bits): Indicates type of encoding currently being used. If sender changes encoding in middle of conference, sender informs the receiver through this payload type field. Payload type 0: PCM mu-law, 64 kbps Payload type 3, GSM, 13 kbps Payload type 7, LPC, 2.4 kbps Payload type 26, Motion JPEG Payload type 31. H.261 Payload type 33, MPEG2 video Sequence Number (16 bits): Increments by one for each RTP packet sent, and may be used to detect packet loss and to restore packet sequence. VoIP protocols 64

65 RTP Header (2) Timestamp field (32 bits long). Reflects the sampling instant of the first byte in the RTP data packet. For audio, timestamp clock typically increments by one for each sampling period (for example, each 125 usecs for a 8 KHz sampling clock) if application generates chunks of 160 encoded samples, then timestamp increases by 160 for each RTP packet when source is active. Timestamp clock continues to increase at constant rate when source is inactive. SSRC field (32 bits long). Identifies the source of the RTP stream. Each stream in a RTP session should have a distinct SSRC. VoIP protocols 65

66 Real-Time Control Protocol (RTCP) Works in conjunction with RTP. Each participant in RTP session periodically transmits RTCP control packets to all other participants. Each RTCP packet contains sender and/or receiver reports report statistics useful to application Statistics include number of packets sent, number of packets lost, interarrival jitter, etc. Feedback can be used to control performance Sender may modify its transmissions based on feedback VoIP protocols 66

67 RTCP - Continued - For an RTP session there is typically a single multicast address; all RTP and RTCP packets belonging to the session use the multicast address. - RTP and RTCP packets are distinguished from each other through the use of distinct port numbers. - To limit traffic, each participant reduces his RTCP traffic as the number of conference participants increases. VoIP protocols 67

68 RTCP Packets Receiver report packets: fraction of packets lost, last sequence number, average interarrival jitter. Sender report packets: SSRC of the RTP stream, the current time, timpstamps of audio and video packets, the number of packets sent, and the number of bytes sent. Source description packets: address of sender, sender's name, SSRC of associated RTP stream. Provide mapping between the SSRC and the user/host name. VoIP protocols 68

69 Synchronization of Streams RTCP can synchronize different media streams within a RTP session. Consider videoconferencing app for which each sender generates one RTP stream for video and one for audio. Timestamps in RTP packets tied to the video and audio sampling clocks not tied to the wallclock time Each RTCP sender-report packet contains (for the most recently generated packet in the associated RTP stream): timestamp of the RTP packet wall-clock time for when packet was created. Receivers can use this association to synchronize the playout of audio and video. VoIP protocols 69

70 RTCP Bandwidth Scaling RTCP attempts to limit its traffic to 5% of the session bandwidth. Example Suppose one sender, sending video at a rate of 2 Mbps. Then RTCP attempts to limit its traffic to 100 kbps. RTCP gives 75% of this rate to the receivers; remaining 25% to the sender The 75 kbps is equally shared among receivers: With R receivers, each receiver gets to send RTCP traffic at 75/R kbps. Sender gets to send RTCP traffic at 25 kbps. Participant determines RTCP packet transmission period by calculating avg RTCP packet size (across the entire session) and dividing by allocated rate. VoIP protocols 70

71 RTP Payload types Payload types (PT) for standard audio and video encodings (Adapted from Tables 4 and 5 of RFC3551) VoIP protocols 71

72 Some RTP payload usage/format RFC 2029 RTP Payload Format of Sun's CellB Video Encoding. RFC 2032 RTP Payload Format for H.261 Video Streams. RFC 2035 RTP Payload Format for JPEG-compressed Video. RFC 2038 RTP Payload Format for MPEG1/MPEG2 Video. RFC 2190 RTP Payload Format for H.263 Video Streams. RFC 2198 RTP Payload for Redundant Audio Data. RFC 2250 RTP Payload Format for MPEG1/MPEG2 Video. RFC 2343 RTP Payload Format for Bundled MPEG. RFC 2429 RTP Payload Format for the 1998 Version of ITU-T Rec. H RFC 2431 RTP Payload Format for BT.656 Video Encoding. RFC 2435 RTP Payload Format for JPEG-compressed Video. RFC 2658 RTP Payload Format for PureVoice(tm) Audio. RFC 2733 An RTP Payload Format for Generic Forward Error Correction. RFC 2793 RTP Payload for Text Conversation. RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony... RFC 3016 RTP Payload Format for MPEG-4 Audio/Visual Streams. RFC 3047 RTP Payload Format for ITU-T Recommendation G RFC 3119 A More Loss-Tolerant RTP Payload Format for MP3 Audio. RFC 3189 RTP Payload Format for DV (IEC 61834) Video. RFC 3190 RTP Payload Format for 12-bit DAT Audio and 20- and 24-bit... RFC 3389 Real-time Transport Protocol (RTP) Payload for Comfort Noise VoIP protocols 72

73 DTMF (Dual Tone Multi Frequency) Touch Tones encode digits 0-9, A-D, *, # as sounds containing two different audio tones Low frequency indicates row, high indicates column Machine readable devices for encoding and decoding (tones back into numbers) are readily available VoIP protocols 73

74 Reference B. Li, M. Hamdi, D. Jiang, X.-R. Cao, and Y. Hou, QoS enabled voice support in the next generation internet: issues, existing approaches and challenges, IEEE Communications Magazine, vol. 38, no. 4, pp , Apr F. Cuervo, N. Greene, A. Rayhan, C. Huitema, B.Rosen, Marconi,J. Segers, Megaco Protocol Version 1.0, Internet draft, Work in progress, Internet Engineering Task Force (RFC3015) C. Groves, M. Pantaleo, LM Ericsson,T. Anderson,T. Taylor, Gateway Control Protocol Version 1, Internet draft, Work in progress, Internet Engineering Task Force (RFC3525) RADVision. Inc, Implementing Gateway Control Protocols, white paper, January 27, 2002 Sean Christensen, Voice over IP solutions, white paper, Juniper Networks, Inc. VoIP protocols 74