Transfer and Control Protocols H.261. Standards of ITU

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1 Transfer and Control Protocols Chapter 2: Basics Chapter 3: Multimedia Systems Communication Aspects and Services Multimedia Applications and Communication Multimedia Transfer and Control Protocols Quality of Service and Resource Management Synchronization Multimedia Operating Systems Chapter 4: Multimedia Systems Storage Aspects Chapter 5: Multimedia Usage 3.2: Transfer and Control Protocols The H.x Protocols Session Initiation Protocol SIP Streaming Multimedia Data Transport Protocols: RTP and RTCP VoIPExample A main protocol family is the H.x standards by the ITU H.261 and H.263 define video coding for video conferences, similar to MPEG H.323 is a control protocol for cooperative computing (session management) Developed by ITU, driven by telecommunication needs Alternative for session management: Session Initiation Protocol (SIP) Only one protocol, not a protocol family Developed by IETF: integrated with the Internet Additionally: RTP/RTCP as transfer protocols H.x and SIP both are not defining transport subsystems RTP as an addition to UDP Page 1 Page 2 Standards of ITU H.261 The ITU has standardized everything needed in cooperative computing: G.711, G.722, G.723, G.728, G.729 for audio coding with kbit/s H.261, H.263, H.264, for video coding similar to MPEG H.245 for controlling media streams H.450 for negotiation of communication resources H.235 for authentication and ciphering H for connection setup and termination, packetizing of data streams, signaling, H.323 for controlling and coordination and several more, e.g. T.x for data transfer User Interface Audio Video Configuration Audio Codecs G.711 G.722 H.323 Video Codecs H.261 H.263 H Layer Network Interface H.245 H.450 H.235 For video conferencing systems, coding/decoding in real-time is required H.261 was designed for ISDN It is a video codec for audiovisual services at p 64 Kbit/s (p = 1, 2, 3,..., 30, referring to ISDN) H.261 can be denoted as px64 Real-time processing requirement of encoding and decoding considered in this standard: maximum signal delay 150 ms (this requirement is a kind of limitation concerning coding and decoding procedures) Page 3 Page 4

2 Properties of H.261 Image format precisely defined Image refresh frequency at input: frames/sec Image encoded as luminance signal Y and chrominance difference signals C B, C R (according to a 4:1:1 subsampling scheme, later adopted by MPEG) 3 basic information from which the full color may be constructed 2 resolution formats (each with 4:3 aspect ratio): Common Intermediate Format (CIF) luminance 288 lines 352 pixels (8 bit per pixel) chrominance Quarter-CIF (QCIF) luminance chrominance QCIF is mandatory for all H.261 implementations, CIF is optional Image Preparation Image is subdivided into blocks of size 8 lines 8 pixels (luminance & chrominance) Macro blocks consists of 4 luminance blocks and 2 corresponding chrominance blocks Group of blocks (GOB) = combination of 33 macro blocks QCIF image consists of 3 GOBs (= = pixels for luminance), CIF image of 12 GOBs (= = pixels for luminance) Note: color difference samples placed such that their block boundaries coincide with luminance block boundaries: Luminance sample Chrominance sample Block edge Block Page 5 Page 6 Data Amount Interframe Coding Uncompressed QCIF: Data rate = frames/sec ( ) bytes/frame Mbit/sec Uncompressed CIF: Data rate Mbit/sec (= Mbit/sec) Compressed QCIF: Needs only 10 frames per second (instead of frames/sec), i.e. three times less: Mbit/sec are required Compression ratio in order to transmit uncompressed Mbit/sec via a 64 Kbit/sec line: 64/ : 47.5 This is possible for today's technology, but only for slow moving pictures Compressed CIF would need 4-6 ISDN B-channels for the same purpose Coding Algorithms: 2 different methods of coding (choice up to the coding control strategy): Intraframe coding: like in JPEG with DCT, quantization and entropy encoding Interframe coding: use of information from previous frame (P frames in MPEG) Prediction for each macro block by motion compensation and spatial filter Motion compensation (similar to MPEG): Comparison of macro blocks from previous and current image motion vector defined by relative position of previous and current macro block One motion vector per macro block, used for all luminance and chrominance blocks Simple implementations just compare previous and actual macro blocks at the same position. In such case, the motion vector is a zero vector Optionally (but rarely used) a low pass filter between DCT and entropy encoding can be used for deleting any remaining high-frequency noise Linear quantization (step size adjusted according to data amount in transformation buffer) Constant data rate at encoder output enforced Quality of encoded video data depends on image contents and motion within scene Page 7 Page 8

3 Data Streams in H.261 H.263 and H.264 Characteristics of Data Stream for H.261: Data stream produced by H.261 has a hierarchical structure several layers, like in MPEG (bottom layer containing compressed picture) Data stream includes information for error correction (18 parity bits for 492 data bits) 5-bit image number as temporal reference for each image Freezing of image which was shown last is possible by an application command; this allows the application at the decoding station to stop and start a video scene in a convenient way Switching between still images and moving images possible (by encoder command!) Conclusion: Suited for applications which do not require too much quality and where the content doesn t move too fast (video conferencing) H.263 is similar to H.261, but defines 5 image formats (sub-qcif, QCIF, CIF, 4CIF, 16CIF) error correction is optional consideration of GOBs and Slices like in MPEG H.264 additionally allows variable block size (16 16, 8 16, 16 8, 8 8) uses a very simple 4 4 transform instead of DCT (astonishingly with negligible loss in quality!) allows to use any frame as a reference for prediction also allows bi-directional prediction (B-frames) allows mixed frames slices of one frame can be coded independently as I-slices, P- slices and B-slices! Page 9 Page 10 H.323 The ITU Family and the OSI Reference Model A video conference is not only transferring video Audio transmission (G.7xx), synchronization with video stream Exchange of configuration data, signalling (H.225, H.245) Whiteboard, chat, application sharing, data, fax (T.x) Transport subsystem (TCP, UDP, RTP, RTCP) H.323 for coordination Not only client terminals (telephones, video phones, NetMeeting, ) speak H.323, but also other system components: Gatekeeper: address translation (phone numbers to IP addresses), admission control and bandwidth management for multipoint connections, call authorization, call signal routing Gateway: integration with other voice networks Multipoint control unit (MCU): coordinates several terminals taking part in a conference Proxy: e.g. used to pass a firewall Transfer of multimedia data uses UDP, transfer of control information uses TCP H.323 is an umbrella standard comprising all the other functionality H.323 APPLICATION PRESENTATION SESSION G.711 G.728 G.722 G.729 Audio Signal G RTCP RAS RTP TRANSPORT UDP NETWORK DATA LINK PHYSICAL H.235 Control H.245 H.225 Video Signal Data H.261 H.263 T.127 Supplementary Services H H H TCP T.126 T.124 T.125/ T.122 X Page 11 Page 12

4 H.323 Network Components H.323 Components and Signaling H.225/RAS messages H.225/RAS messages H.225/Q.931 (optional) Gatekeeper H.225/Q.931 (optional) H.245 messages (optional) H.245 messages (optional) H.225/Q.931 messages over call signaling channel POTS Terminal H.245 messages over call control channel Gateway H.323 terminal can be workstations as well as more specalized end systems, e.g. IP phones The gateway enables an integration with existing systems like ISDN or older POTS (Plain Old Telephony System) Page 13 H.245 A protocol for capabilities advertisement, media channel establishment and conference control. H Call Control Q.931 A protocol for call control and call setup. RAS Registration, admission and status protocol used for communicating between an H.323 endpoint and a gatekeeper. Page 14 Process for Establishing Communication Establishing communication using H.323 occurs in five steps: 1. Call setup 2. Initial communication and capabilities exchange 3. Audio/video communication establishment 4. Call services 5. Call termination Page 15 Simplified H.323 Call Setup Both endpoints have previously registered with the gatekeeper Terminal A initiate the call to the gatekeeper The gatekeeper provides information for Terminal A to contact Terminal B Terminal A sends a SETUP message to Terminal B Terminal B responds with a Call Proceeding message and also contacts the gatekeeper for permission Terminal B sends a Alerting and Connect message Terminal B and A exchange H.245 messages to determine master/slave, terminal capabilities, and open logical channels The two terminals establish RTP media paths for data transmission Terminal A Gatekeeper Terminal B 1. ARQ 2. ACF 3. SETUP 4. Call Proceeding 5. ARQ 6. ACF 7.Alerting 8.Connect H.245 Messages RTP Media Path RAS messages Call Signaling Messages Note: This diagram only illustrates a simple pointto-point call setup where call signaling is not routed to the gatekeeper. Refer to the H.323 recommendation for more call setup scenarios. Page 16

5 Session Initiation Protocol SIP Defined by 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 SIP is an application layer signaling protocol that defines initiation, modification and termination of interactive multimedia communication sessions between multiple users Bases upon HTTP concepts (message syntax, SIP URLs, responses, ) SIP Services Setting up a call Provides mechanisms for caller to let callee know he 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 Page 17 Page 18 Setting up a Call to a known IP Address Call Setup µ Alice s SIP invite message indicates her port number & IP address. Indicates encoding that Alice prefers to receive (PCM µlaw) Bob s 200 OK message indicates his port number, IP address & preferred encoding (GSM) SIP messages can be sent over TCP or UDP; here sent over RTP/UDP Default SIP port number is 506. Codec negotiation Suppose Bob doesn t have PCM µlaw 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. Page 19 Page 20

6 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 506. Alice specifies in Via: header that SIP client sends and receives SIP messages over UDP Name Translation and User Location 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 SIP redirect server SIP location server Page 21 Page 22 SIP Distributed Architecture SIP Registrar User Agent SIP Components Location Server Proxy Server Redirect Server Proxy Server Registrar Server Gateway User Agent Client (UAC) An entity that initiates a call User Agent Server (UAS) An entity that receives a call PSTN When Bob starts SIP client, the client sends SIP REGISTER message to Bob s registrar server Register Message: REGISTER sip:domain.com SIP/2.0 Via: SIP/2.0/UDP From: sip:[email protected] To: sip:[email protected] Expires: 3600 Page 23 Page 24

7 SIP Proxy Example 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 Interprets, rewrites or translates a request message before forwarding it Note: proxy is analogous to local DNS server Caller [email protected] places a call to [email protected] (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 [email protected] (4) umass proxy sends INVITE to eurecom registrar (5) eurecom regristrar forwards INVITE to , which is running keith s SIP client SIP proxy umass.edu 1 8 SIP client SIP registrar upenn.edu (6-8) SIP response sent back (9) Messages sent directly between clients 4 7 SIP registrar eurecom.fr 6 5 SIP client Note: also a SIP ack message, which is not shown 9 Page 25 Page 26 Comparison with H.323 Transport Subsystem H.323 is a complete, vertically integrated suite of protocols for multimedia conferencing: signaling, registration, admission control, transport and codecs SIP is a single component. Works with RTP, but does not mandate it. Can be combined with other protocols and services. H.323 comes from the ITU (telephony) SIP comes from IETF: Borrows much of its concepts from HTTP. SIP has a Web flavor, whereas H.323 has a telephony flavor How to transfer multimedia data in the Internet? TCP/UDP/IP: best-effort service No guarantees on delay, loss Today s Internet multimedia applications use application-level techniques to mitigate (as best as possible) effects of delay and loss E.g. streamed stored multimedia Application-level streaming techniques for making the best out of best effort service: Client side buffering Use of UDP versus TCP Multiple encodings of multimedia H.323 is complex SIP uses the KISS principle: Keep it simple and stupid But: what protocols on lower layers are suitable to support such application-level streaming? Page 27 Page 28

8 Internet Multimedia: Simplest Approach Internet Multimedia: Streaming Approach First: how can application level streaming be realized? Audio or video stored in files Files are transferred as HTTP object Received in entirety at client Then passed to player Audio and video are not really streamed: Long delays until playout! Browser GETs metafile with server contact information Browser launches player, passing metafile Player contacts server Server streams audio/video to player Page 29 Page 30 Streaming from a Streaming Server Streaming Multimedia: Client Buffering variable fill rate x(t) constant drain rate d Separation of web server and streaming This architecture allows for non-http protocol between server and media player Can also use UDP instead of TCP buffered video In streaming, data can arrive with variable rate by network delay and jitter Thus: client-side buffering for playout delay for compensation of these problems Page 31 Page 32

9 Streaming Multimedia Solution: RTSP What transport protocol to use for such an approach? UDP Server sends at rate appropriate for client (oblivious to network congestion!) Often send rate = encoding rate = constant rate Then: fill rate = constant rate - packet loss Short playout delay (2-5 seconds) to compensate for network jitter Error recovery: if time permits TCP Send at maximum possible rate under TCP Fill rate fluctuates due to TCP congestion control Larger playout delay: smooth TCP delivery rate HTTP/TCP passes more easily through firewalls HTTP Does not target multimedia content No commands for fast forward, etc. Real-time Streaming Protocol RTSP Client-server application layer protocol For user to control display: rewind, fast forward, pause, resume, repositioning, etc What it doesn t do: Does not define how audio/video is encapsulated for streaming over network Does not restrict how streamed media is transported; it can be transported over UDP or TCP Does not specify how the media player buffers audio/video Page 33 Page 34 RTSP: Out of Band Control RTSP Example FTP uses an out-of-band control channel: A file is transferred over one TCP connection Control information (directory changes, file deletion, file renaming, etc.) is sent over a separate TCP connection The out-of-band and in-band channels use different port numbers RTSP messages are also sent out-of-band: RTSP control messages use different port numbers than the media stream (Port 554): out-of-band The media stream is considered in-band Scenario: Metafile communicated to web browser Browser launches player Player sets up an RTSP control connection and a data connection to streaming server Page 35 Page 36

10 Metafile Example RTSP Exchange Example <title>twister</title> <session> <group language=en lipsync> <switch> <track type=audio e="pcmu/8000/1" src = "rtsp://audio.example.com/twister/audio.en/lofi"> <track type=audio e="dvi4/16000/2" pt="90 DVI4/8000/1" src="rtsp://audio.example.com/twister/audio.en/hifi"> </switch> <track type="video/jpeg" src="rtsp://video.example.com/twister/video"> </group> </session> C: SETUP rtsp://audio.example.com/twister/audio RTSP/1.0 Transport: rtp/udp; compression; port=3056; mode=play S: RTSP/ OK Session 4231 C: PLAY rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Session: 4231 Range: npt=0- C: PAUSE rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Session: 4231 Range: npt=37 C: TEARDOWN rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Session: 4231 S: OK Page 37 Page 38 Example: Internet Phone Jitter Introduce Internet Phone by way of an example Speaker s audio: alternating talk spurts, silent periods 64 kbit/s during talk spurt Packets are generated only during talk spurts 20 msec chunks at 64 kbit/sec: 160 bytes data Application-layer header added to each chunk Chunk and header are encapsulated into a UDP segment. Application sends UDP segments into socket every 20 msec during talkspurt Required: Network loss: IP datagram lost due to network congestion (router buffer overflow) Delay loss: IP datagram arrives too late for playout at receiver Delays: processing, queueing in network; end-system (sender, receiver) delays Typical maximum tolerable delay: 400 ms Loss tolerance: depending on voice encoding, losses concealed, packet loss rates between 1% and 10% can be tolerated Cumulative data constant bit rate transmission variable network delay (jitter) client playout delay client reception buffered data constant bit rate playout at client Consider the end-to-end delays of two consecutive packets: difference can be more or less than 20 msec time Page 39 Page 40

11 Internet Phone: Fixed Playout Delay Receiver attempts to playout each chunk exactly q msecs after chunk was generated chunk has timestamp t: play out chunk at t + q chunk arrives after t + q: data arrives too late for playout, data lost Tradeoff for q: large q: less packet loss small q: better interactive experience r: receiving of first packet p: first playout schedule p : second playout schedule Adaptive Playout Delay Goal: minimize playout delay, keeping late loss rate low Approach: adaptive playout delay adjustment: Estimate network delay, adjust playout delay at beginning of each talk spurt Silent periods compressed and elongated Chunks still played out every 20 msec during talk spurt. ti = timestamp of the ith packet ri = the time packet i is received by receiver pi = the time packet i is played at receiver * d i = ri ti = network delay for ith packet d = estimate of average network delay after receiving ith packet i 20 msec Dynamic estimate of average delay at receiver: d = (1 u) d + u d * i i 1 i where u is a fixed constant (e.g., u =.01) Page 41 Page 42 Adaptive Playout Delay Recovery from Packet Loss Also useful to estimate the average deviation of the delay v i (jitter): v = (1 u) v + u d d * i i 1 i i The estimates d i and v i are calculated for every received packet, although they are only used at the beginning of a talk spurt. For first packet in talk spurt, playout time is: pi = ti + di + Kvi where K is a positive constant. Remaining packets in talkspurt are played out periodically Forward error correction (FEC): simple scheme For every group of n chunks create a redundant chunk by exclusive OR-ing the n original chunks Send out n+1 chunks, increasing the bandwidth by factor 1/n. Can reconstruct the original n chunks if there is at most one lost chunk from the n+1 chunks Playout delay needs to be fixed to the time to receive all n+1 packets Tradeoff: increase n, less bandwidth waste increase n, longer playout delay increase n, higher probability that 2 or more chunks will be lost Page 43 Page 44

12 Recovery from Packet Loss Recovery from Packet Loss Other FEC scheme: Piggyback lower quality stream Send lower resolution audio stream as the redundant information For example, nominal stream PCM at 64 kbps and redundant stream GSM at 13 kbps lower quality Whenever there is non-consecutive loss, the receiver can conceal the loss Can also append (n-1)st and (n-2)nd low-bit rate chunk Interleaving Chunks are broken up into smaller units For example, 45 msec units per chunk Packet contains small units from different chunks If packet is lost, still have most of every chunk Has no redundancy overhead But adds to playout delay Page 45 Page 46 Summary: Internet Multimedia: Bag of Tricks Real-Time Protocol (RTP) Use UDP to avoid TCP congestion control (delays) for time-sensitive traffic Client-side adaptive playout delay to compensate for network delay Server side matches stream bandwidth to available client-to-server path bandwidth Chose among pre-encoded stream rates Dynamic server encoding rate Error recovery (on top of UDP) FEC, interleaving Retransmissions, time permitting Conceal errors: repeat nearby data Provide a standardized transport protocol which supports such tricks: RTP RTSP still would have to use the unreliable UDP or the slow TCP better define a new transport protocol for combining speed with reliability: Real-Time Transport Protocol (RTP) RTP specifies a packet structure for packets carrying audio and video data 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 Page 47 Page 48

13 RTP runs on Top of UDP RTP Header RTP libraries provide a transport-layer interface that extend UDP: Port numbers, IP addresses Payload type identification Packet sequence numbering Time-stamping Transport Layer Page 49 Ver.: Version number of the RTP protocol in use P: packet size was padded to a multiple of 32 bit X: an extension header is used CC: indicates the number of sources M: User-specific mark. Can e.g. mark the beginning of a word on an audio channel. Contributing Source Identifier: used by mixers in the studio. The mixed flows are listed here. Page 50 RTP Header RTP Header Payload Type (7 bits) Indicates type of encoding currently being used. If the sender changes encoding in middle of transmission, it informs the receiver through this payload type field Payload type 0: PCM µ-law, 64 kbps Payload type 3, GSM, 13 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 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 µsecs 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. The timestamp gives the receiver the relative time (with respect to the first data) when to playout the data Synchronization Source Identifier field (32 bits long) Identifies the source of the RTP stream Each stream in a RTP session should have a distinct identifier Page 51 Page 52

14 RTP and QoS Real-Time Control Protocol (RTCP) RTP only adds some information to the UDP header needed for kind of reliability 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. Usage of (and reaction to) the information in the RTP header are left over to the application 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 To limit traffic, each participant reduces his RTCP traffic as the number of conference participants increases Page 53 Page 54 RTCP Application Example: Voice over IP (VoIP) RTCP controls the data flow: Feedback to the sender about QoS on receiver side Data losses, delay and jitter are reported Note: RTCP does not provide corrective actions - this is left over to the application Sender Application RTP / RTCP UDP IP Receiver Application RTP / RTCP UDP IP Telephony using an IP network with standardized protocols: VoIP Transferring speech and signaling information Not only internally in a IP network, also integration with normal telephony systems IP phones IP network (Internet/Intranet) VoIP Gateway IP terminal IP addresses and virtual phone numbers RTCP RTP RTP RTP RTP RTP RTP RTCP RTCP RTCP ISDN phone Telecommunication network Phone numbers Page 55 Page 56

15 VoIP-based Telephony System Realization with H.323 H.323 zone Branch ISDN PTSS ISDN Company central ISDN PTSS H.323 terminal Network without QoS guarantees Gatekeeper MCU H.323 terminal VoIP- Gateway Teleworking PCs VoIP- Gateway Gateway R IP network (Internet) R POTS H.324 ISDN H.320 ATM network H.321 R = Router PTSS = Private Telecommunications Switching System Page 57 H.323 gives us all functionality we need to realize an IP-based telephony integrated with conventional solutions Page 58 Call Setup PTSS ISDN VoIP Future GK: Gatekeeper IP phone Ringing Intranet GK Phone nr => IP-Adr.? IP[TCP[SETUP[...]]] IP[TCP[Alerting [...]]] VoIP Gateway SETUP [...] Call Proceeding Alerting ISDN phone Call is initiated Dialing tone At the moment, VoIP products based on H.323 are most popular But: complex, and telecommunication networks tend to converge with IP networks Use protocols better integrated with the IP world SIP together with a MGCP (Media Gateway Control Protocol) gets more and more significance Better integration with web applications SIP seems to be the multimedia signaling protocol for the future Pick up receiver IP[TCP[Connect[...]]] RTP channel Connect B channel Connection Still a problem: quality of an IP transmission; how to improve QoS in the Internet? IP[UDP[RTP[Voice]]] Voice Page 59 Page 60