Multimedia and the Internet More and more multimedia streaming applications in the Internet: Video on Demand IP telephony Internet radio Teleconferencing Interactive Games Virtual/augmented Reality Tele learning Requirements for data transfer Low end-to-end delay Almost no delay variation (jitter) Need for special streaming protocols in the application layer Communication Networks -11. The Internet 602 Multimedia Clients Tasks: Decompression of the coded multimedia stream on the fly Buffering for mitigation of jitter effect Error correction Forward error correction Explicit retransmission only in exceptional cases Interpolation of lost data Provision of a graphical user interface Different Types: Stand alone applications, e.g. VLC Player, MS Media Player Plug-Ins for web browsers, e.g. Flash Communication Networks -11. The Internet 603 Prof. Jochen Seitz 1
Options for Transporting a Multimedia Stream Transport stream at a constant data rate using UDP and prompt presentation at client side E.g. audio stream with 13 kb/s As soon as audio samples have been received, they are decompressed and played back Transport stream at a constant data rate using UDP and delayed presentation at client side Client has to buffer the stream for a short time (few seconds) in order to eliminate jitter Transport stream using TCP and delayed presentation at client side Better quality due to error correction mechanisms of TCP Danger of empty buffer because of TCP s slow start Communication Networks -11. The Internet 604 9.7 Multimedia and the Internet Real Time Streaming Protocol (RTSP) Control of multimedia stream (comparable to a blue ray player) : Pause (Re-)Positioning of the stream Fast forward or rewind with playback... Protocol to transmit this control information: Real Time Streaming Protocol (RTSP) RTSP is not responsible for Media compression / decompression Packetizing the media stream Transporting the media stream Buffering the data Communication Networks -11. The Internet 605 Prof. Jochen Seitz 2
RTSP Message Sequence Chart Web Browser HTTP Get Session Description Web Server SETUP Media Player PLAY RTP audio RTP video RTCP PAUSE Media Server TEARDOWN Communication Networks -11. The Internet 606 RTSP Commands DESCRIBE retrieval of the description of a media object from a server GET_PARAMETER retrieval of a given parameter value, e.g. media property OPTIONS PLAY PLAY_NOTIFY PAUSE REDIRECT SETUP requesting the methods that are supported by the RTSP agent / that are allowed for the specified resource starting playback at a given position/play time notification of the client about an asynchronous event for a running session, e.g. scale change immediately interrupt the stream delivery inform a client that the service provided will be terminated and where a corresponding service can be provided instead establishing/modifying an RTSP session between client and server SET_PARAMETER setting of the value of a parameter or a set of parameters for a presentation or stream TEARDOWN stop the stream and free the resources associated with it Communication Networks -11. The Internet 607 Prof. Jochen Seitz 3
RTSP Characteristics Out-Of-Band control protocol Port number 544 RTSP uses TCP or UDP Reliability If TCP is used, every RTSP command will be sent once If UDP is used, the RTSP command will be repeated after an estimated round trip time in case the command does not cause the expected output Commands are numbered to avoid double execution Communication Networks -11. The Internet 608 A Transport Protocol for Real-Time Applications (RTP) Motivation: Standardized packet structure with fields for Audio data Video data Sequence numbers Time stamps Usage of UDP to transmit media streams RTP is located between application providing the multimedia streams and transport layer Transmission of existing standardized media formats for audio: PCM(A or U), GSM, for Video: H.261, JPEG,. Important: RTP cannot give guarantees for the timely transport of data or other QoS requirements Communication Networks -11. The Internet 609 Prof. Jochen Seitz 4
RTP in the Internet Reference Model Internet Reference Model Application RTP Transport (UDP) RTP also supports multicast and multipeer RTP Session Internet (IP) Network-to-Host Communication Networks -11. The Internet 610 RTP Packet Header Payload Type Sequence Number Timestamp Synchronization Source ID Miscellaneous Fields 32 bit, identifies the source of the RTP stream (not an IP address, but a random number generated at the beginning of the transmission) 32 bit, derived from the sender sampling clock relating to the first byte in the packet 16 bit, increased by 1 in every new packet to detect and eventually retransmit missing packets 7 bit, defines the coding of the stream Communication Networks -11. The Internet 611 Prof. Jochen Seitz 5
RTP Payload Types Payload Type Audio Format Sampling Rate Data Rate 0 PCM m-law 8 khz 64 kb/s 1 1016 8 khz 4,8 kb/s 3 GSM 8 khz 13 kb/s 7 Linear Predictive Coding 8 khz 2,4 kb/s 9 G.722 8 khz 48 64 kb/s 14 MPEG Audio 90 khz 15 G.728 8 khz 16 kb/s Payload Type Video Format 26 Motion JPEG 31 H.261 32 MPEG1 Video 33 MPEG2 Video Communication Networks -11. The Internet 612 RTP Control Protocol (RTCP) Control protocol specified for RTP RTCP packets periodically transmitted by sender and receiver(s) to provide information to the application about the quality of the transmisison: Number of sent packets Number of lost packets Estimation of jitter RTCP does not specify what the application how the application hast to react Communication Networks -11. The Internet 613 Prof. Jochen Seitz 6
RTCP Packet Types Receiver Report (for each RTP stream) Reception statistics from participants: Synchronization Source ID Ratio of lost packets Least received sequence number Mean deviation of interarrival times Sender Report Transmission and reception statistics from active senders Synchronization Source ID Timestamp of last sent packet Local time at sender Number of sent packets and bytes Reports allow synchronization of different streams Communication Networks -11. The Internet 614 RTCP Data Rate Scaling Problem: Control traffic might exceed data traffic! Solution: Adaptation of RTCP packet rate Goal: Limitation of RTCP traffic to 5% of the overall traffic in a session Multicast example: Overall traffic : 2 Mb/s RTCP traffic: 100 kb/s Distribution of RTCP traffic (for example): 25 % for sender: 25 kb/s (single sender) 75 % for all R receivers: 75 kb/s 1 Period for sender report: T RTCP Packet Size 0,25 0,05 2Mb / s R Period for receiver report: T RTCP Packet Size 0,75 0,05 2Mb / s Communication Networks -11. The Internet 615 Prof. Jochen Seitz 7
9.8 Voice over IP Voice over IP (VoIP) PC to PC: IP-based Network (Intra- or Internet) PC to traditional telephone: IP-based Network (Intra- or Internet) VoIP Gateway PSTN ISDN PBX Communication Networks -11. The Internet 616 9.8 Voice over IP Protocols for VoIP Protocols for the transport of digital voice samples: RTP, RTCP Protocols for the transport of control information (signaling): H.323, Session Initiation Protocol (SIP) Protocols for the integration of traditional telecommunication systems into VoIP (Media Gateway Control Protocols) : MGCP, Megaco Communication Networks -11. The Internet 617 Prof. Jochen Seitz 8
9.8 Voice over IP Integration of IP and Telephony by H.323 IP-Network POTS Data Optimized for data transmission Efficient usage of capacity/bandwidth Voice Video Optimized for voice/ video communication Isochronous transmission H.323 interconnects these different worlds Communication Networks -11. The Internet 618 9.8 Voice over IP H.323 H.323 is a standard for real time audio and video conferencing in the Internet H.323 additionally specifies the interoperability between Internet end devices and traditional telephones H.323 defines an application context for RTP RTP is one main component of H.323 H.323 devices are for example End Points Stand-alone devices like web phones or web TV sets PC applications like software for Internet video conferencing) Gateways For the transition into the POTS Gatekeepers For address translation, authentication, accounting, bandwidth management... Communication Networks -11. The Internet 619 Prof. Jochen Seitz 9
9.8 Voice over IP H.323 Basic Devices H.323 Terminal Multipoint H.323 Terminal Control Unit MCU H.323 Zone Gatekeeper VoIP- Gateway POTS ISDN H.324 H.320 B-ISDN H.321 Communication Networks -11. The Internet 620 9.8 Voice over IP H.323 Protocol Architecture Audio Applications Video Applications System Control G.711 G.722 G.729 etc. RTP RTCP H.261 H.263 etc. RAS Channel H.255 Call Signaling Channel Q.931 Call Control Channel H.245 H.323 UDP TCP IP Communication Networks -11. The Internet 621 Prof. Jochen Seitz 10
9.8 Voice over IP H.323 Channels and Sessions An H.323 session comprises different H.323 channels several simultaneous RTP streams for each RTP stream, there usually os a channel for sending and a channel for receiving one RTCP channels (as described before) one H.245 control channel for opening /closing channels negotiating channel characteristics (e.g. which codec should be used) one Q.931 channel for traditional telephony (calling a telephone number, ) Communication Networks -11. The Internet 622 9.7 Voice over IP TCP/IP and VoIP Protocols Signaling Audio / Video H.323-SIG SIP RTP RTCP TCP UDP Internet Protocol Network to Host Layer Communication Networks -11. The Internet 623 Prof. Jochen Seitz 11
9.7 Voice over IP Session Initiation Protocol (SIP) Protocol for signaling and controlling multimedia communication sessions Comparable to H.323 Less complex Easier to implement Based on client/server-model Supports user mobility User registers to SIP registrar with authentication Session Description Protocol (SDP) forming the payload of SIP messages for media type and parameter negotiation media setup Communication Networks -11. The Internet 624 9.7 Voice over IP SIP Message Sequence Chart Romeo@montague.de Julia@capulet.de INVITE Julia@capulet.de 180 Ringing 200 OK ACK Julia@capulet.de VoIP - Session BYE Romeo@montague.de 200 OK Communication Networks -11. The Internet 625 Prof. Jochen Seitz 12
9.7 Voice over IP SIP Interoperability with ISDN INVITE 100 Trying 180 Ringing 200 OK VoIP- Gateway SETUP ALERT CONNECT VoIP Session B Channel BYE 200 OK DISCONNECT RELEASE RELEASE ACKNOWLEDGE Communication Networks -11. The Internet 626 9.7 Voice over IP Integration of Traditional Telecommunication Systems with VoIP Different Gateways for the integration of traditional telecommunication systems into VoIP networks Media Gateways Protocols to control these Media Gateways MGCP (Media Gateway Control Protocol) Megaco Traditional telephones can be used in VoIP systems Communication Networks -11. The Internet 627 Prof. Jochen Seitz 13
9.7 Voice over IP Media Gateways: Types and Application Trunking Gateway: VoIP..... Telephone network Residential Gateway: VoIP..... Access Gateway: VoIP..... PBX Communication Networks -11. The Internet 628 9.7 Voice over IP The Role of VoIP VoIP simply changed the transport of digitalized voice Packet-switched instead of circuit-switched Higher resource efficiency The process for making a phone call is very similar to the traditional telephony In Germany, the German Telekom will stop the ISDN service in 2018 and fully replace it with VoIP Nevertheless, some disadvantages exist: No QoS guarantees No emergency supply Communication Networks -11. The Internet 629 Prof. Jochen Seitz 14
References Requests for Comments Rosenberg, Jonathan; Schulzrinne, Henning; Camarillo, Gonzalo; Johnston, Alan; Peterson, Jon; Sparks, Robert et al. (2002): SIP. Session Initiation Protocol. (RFC 3261). Andreasen, Flemmin; Foster, Bill (2002): Media Gateway Control Protocol (MGCP) Version 1.0. (RFC 3435). Schulzrinne, Henning; Casner, Stephen L.; Frederick, Ron; Jacobson, Van (2003): RTP. A Transport Protocol for Real-Time Applications. (RFC 3550). Schulzrinne, Henning; Casner, Stephen L. (2003): RTP Profile for Audio and Video Conferences with Minimal Control. (RFC 3551). Schulzrinne, Henning; Agboh, Charles (2005): Session Initiation Protocol (SIP)-H.323 Interworking Requirements. (RFC 4123). Handley, Mark; Jacobson, Van; Perkins, Colin (2006): SDP. Session Description Protocol. (RFC 4566). Groves, Christian; Lin, Yangbo (2009): H.248/MEGACO Registration Procedures. (RFC 5615). Ott, Joerg; Perkins, Colin (2010): Guidelines for Extending the RTP Control Protocol (RTCP). (RFC 5968). Schulzrinne, Henning; Rao, Anup; Lanphier, Rob; Westerlund, Magnus; Stiemerling, Martin (2016): Real-Time Streaming Protocol Version 2.0. RFC 7826. Communication Networks -11. The Internet 630 References References Hartpence, Bruce (2013): Packet Guide to Voice over IP. Sebastopol, CA: O'Reilly. Perkins, Colin (2012): RTP. Audio and Video for the Internet. Boston: Addison- Wesley. Simpson, Wes (2008): Video over IP: IPTV, Internet Video, H.264, P2P, Web TV, and Streaming. A Complete Guide to Understanding the Technology. 2nd edition. Amsterdam: Elsevier. Sun, Lingfen; Mkwawa, Is-Haka; Jammeh, Emmanuel; Ifeachor, Emmanuel (2013): Guide to Voice and Video over IP. For Fixed and Mobile Networks. London: Springer London; Imprint: Springer (Computer Communications and Networks). Yeung, Joe (2015): VoIP. A Practical Guide for the Non-Telephone Engineer. Wilmington, NC: Lulu Press, Inc. Communication Networks -11. The Internet 631 Prof. Jochen Seitz 15