TIA-1145-2012 Packet-Switched Video Telephony Stage 1《分组交换食品电话 第1阶段》.pdf

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1、 TIA-1145 January 2012Packet-Switched Video Telephony Stage 1 NOTICE TIA Engineering Standards and Publications are designed to serve the public interest through eliminating misunderstandings between manufacturers and purchasers, facilitating interchangeability and improvement of products, and assis

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16、R PRODUCED OR RENDERED TO COMPLY WITH THE CONTENTS. TIA SHALL NOT BE LIABLE FOR ANY AND ALL DAMAGES, DIRECT OR INDIRECT, ARISING FROM OR RELATING TO ANY USE OF THE CONTENTS CONTAINED HEREIN, INCLUDING WITHOUT LIMITATION ANY AND ALL INDIRECT, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES (INCLUDING DA

17、MAGES FOR LOSS OF BUSINESS, LOSS OF PROFITS, LITIGATION, OR THE LIKE), WHETHER BASED UPON BREACH OF CONTRACT, BREACH OF WARRANTY, TORT (INCLUDING NEGLIGENCE), PRODUCT LIABILITY OR OTHERWISE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGES. THE FOREGOING NEGATION OF DAMAGES IS A FUNDAMENTAL ELEMEN

18、T OF THE USE OF THE CONTENTS HEREOF, AND THESE CONTENTS WOULD NOT BE PUBLISHED BY TIA WITHOUT SUCH LIMITATIONS. Editors Aleksandar Gogic, QUALCOMM, Incorporated, +1 858 651 5386, Revision History Version 1.0 Initial Publication July 21, 2005 Packet-Switched Video Telephony Stage 1 2 1 Contents 1 Co

19、ntents.3 2 Scope5 3 References.5 4 Abbreviations.5 5 Introduction.6 6 Services and Features7 6.1 Service Characteristics.7 6.2 Service Structure 7 7 Service Requirements 8 7.1 System .8 7.2 User Configuration .8 7.3 Service Interaction .9 7.4 Video Codec 9 7.5 Speech Codec10 7.6 Packetization .10 8

20、QoS Requirements10 8.1 General.10 8.2 QoS Attributes.11 8.2.1 Maximum and Average Data Rate (kbps) 11 8.2.2 Maximum Video Frame Rate (fps) 11 8.2.3 Maximum Transfer Delay (sec) 8.2.4 Inter-Media Skew (sec) 11 8.2.5 Frame Error Rate 12 8.2.6 Jitter (sec) 8.3 Control of VT Parameters12 9 Security 13 1

21、0 Accounting .13 11 Call Scenarios (Informative)14 a. Normal VT Call Flow.14 b. Unsuccessful VT Call14 c. Fallback of VT Call14 Packet-Switched Video Telephony Stage 1 3 d. Incoming Voice Call during VT Call14 e. Incoming VT Call during VT Call 14 f. Teardown of VT Call 15 Packet-Switched Video Tele

22、phony Stage 1 4 2 Scope This document defines the functional characteristics and the requirements of the packet-switched Video Telephony (VT) services, sometimes also referred to as Multimedia Conversational Services (MCS). Video Telephony is defined as one-to-one video/speech communication capabili

23、ty. VT (MCS) should be used as a basis for multiparty multimedia conferencing, but this is out of scope for the initial document revision. This initial effort is to be mindful of the continued development in subsequent phases of the following enhancements (without implying any relative priority): nu

24、ll Interoperability support to other video telephony systems; null Capability of multipoint communication. Hence the following two phases in the development of video telephony are defined: Phase 1: Development of basic one-on-one video telephony functionality (Subject to Stage 1 Revision 0) Phase 2:

25、 Enhancement for support of other video telephony systems and multipoint communication (Subject to Stage 1 Revision A or later) 3 References 1 ITU-T Recommendation H.323: “Visual Telephone Systems and Equipment for Local Area Networks which Provide a Non-Guaranteed Quality of Service” 2 ITU-T Recomm

26、endation H.324: “Terminal for Low Bitrate Multimedia Communication” 3 C.S0009-0 v1.0 Speech Service Option Standard for Wideband Spread Spectrum Systems 4 C.S0014-A v1.0 Enhanced Variable Rate Codec, Speech Service Option 3 for Wideband Spread Spectrum Digital Systems 5 C.S0020-A v1.0 High Rate Spee

27、ch Service Option 17 for Wide Band Spread Spectrum Communication Systems 6 C.S0030-0 v3.0 Selectable Mode Vocoder (SMV) Service Option for Wideband Spread Spectrum Communication Systems 7 C.S0052-A v1.0 Source-Controlled Variable-Rate Multimode Wideband Speech Codec (VMR-WB), Service Options 62 and

28、63 for Spread Spectrum Systems 8 IETF RFC 3261: “Session Initiation Protocol” (SIP) 9 IETF RFC 3550: “RTP: A Transport Protocol for Real-Time Applications” (a.k.a. Real Time Protocol) - IETF, July 2003 4 Abbreviations 3G Third Generation system 3GPP2 Third Generation Partnership Project 2 BER Bit Er

29、ror Rate CIF ITU-T Common Intermediate Format (352 pixels x 288 lines) FER Frame Error Rate Packet-Switched Video Telephony Stage 1 5 IETF Internet Engineering Task Force ISO International Standards Organization ITU-T International Telecommunication Union - Telecommunication Sector MCS Multimedia Co

30、nversational Services MDN Mobile Directory Number MOS Mean Opinion Score PSNR Peak Signal-to-Noise Ratio a quality metric for video codec on a lossy transmission channel QCIF ITU-T Quarter Common Intermediate Format (176 pixels x 144 lines) QoS Quality of Service RFC Request for Comments RLP Radio L

31、ink Protocol RTP Real Time Protocol (See RFC 3550) SIP Session Initiation Protocol URL Uniform Resource Locator VT Video Telephony 5 Introduction The potential benefits of communicating via visual media, in addition to speech, have long been recognized, as greatly enhancing the potential for users t

32、o communicate and to convey information. Video telephony consists of ability of a user to simultaneously talk to another party, view video images from the other party, and send to the other party video images captured by local camera, stored in local device, etc. User can stop video communication wi

33、thout interrupting voice conversation. Transmitting a video stream has proven to be a very challenging goal, since it requires significant resources, and therefore can place a heavy burden on the system. Because a video stream contains much more information than voice alone, it demands much higher d

34、ata rate. In order to reduce the data rate, many video codecs are optimized for maximum compression alone, and thus are sensitive to transmission errors. With the development of 3G wireless communications systems, the data rate available to each user can be considerably increased. The available netw

35、ork throughput has reached the threshold where reasonable quality video telephony services can be realized. New developments in packet (IP based) networks as of late, have made video telephony a more viable service due to some of the following: null Standardization of comprehensive QoS capabilities

36、in all variants of cdma20001wireless networks; null Payload flexibility of packet switching and routing, which enables asymmetric and variable rate bearers; 1cdma2000 is the trademark for the technical nomenclature for certain specifications and standards of the Organizational Partners (OPs) of 3GPP

37、2. Geographically (and as of the date of publication), cdma2000 is a registered trademark of the Telecommunications Industry Association (TIA-USA) in the United States. Packet-Switched Video Telephony Stage 1 6 null Optimization of wireline protocols for wireless networks (e.g. protocol header compr

38、ession/removal); null Increase in number of devices (PCs and laptop computers) frequently attached to a network (wide area wireless, WLAN, or wired), in addition to the camera-equipped wireless phones. 6 Services and Features 6.1 Service Characteristics Packet-Switched Video Telephony provides two-w

39、ay transmission of real-time video, speech, combined speech/video, and other services in the wireless communications systems. These services can be used in videophone, multimedia video conferencing, long-distance classroom, video communications in fieldwork, etc. Latency (round-trip delay between th

40、e parties) needs to be kept to a minimum in order to support real-time information exchange between the parties. In general, it is not feasible to recover transmission errors by retransmission techniques, because of the delays involved in requesting retransmission. Occasionally, video and speech qua

41、lity may need to be compromised to meet latency requirements. Video and speech codecs must be robust enough to cope with the possibilities of errors in wireless transmission channels. Intelligent use of error-recovery and error-concealment techniques is desirable to ensure graceful performance degra

42、dation on noisy channels. 6.2 Service Structure A VT terminal supports the capture, transport, and playback of real-time full-duplex speech and video. The VT terminal contains a speech codec that produces good voice quality and a video codec that produces reasonable video quality. The terminal sets

43、up, maintains, and releases video telephony sessions, and transmits and receives the following data flows: null a low data rate packet loss sensitive control flow; null a low data rate and delay sensitive speech flow; and null a medium data rate delay sensitive video flow. Assuming provision of adeq

44、uate network resource (e.g. sufficient radio resource capacity to handle the traffic load), video telephony services can in principle exist alongside other wireless communication services without mutual impact. VT services are symmetric between the parties in terms of available functionality, which

45、includes encoding, transmission, and decoding components. However, to increase flexibility and optimize radio spectrum efficiency, the services can be implemented without requiring symmetric bearers on the forward and the reverse link, as well as to allow the user to add or drop a multimedia compone

46、nt, and the operator to manage access based on the network traffic load, QoS, etc. Video and speech codecs encode the video and speech signals acquired by the video terminal. After packetization, the encoded data is transmitted by the wireless network, and delivered to the other participant of the V

47、T call using the Internet for transport and routing. At the same time, the VT terminal processes incoming video and speech data from the remote participant by de-packetizing them, then decoding this data with corresponding video and speech codecs. The output is then played back at the local Packet-S

48、witched Video Telephony Stage 1 7 video/audio device. Video telephony also includes system control protocols for setting up calls between parties, exchanging and negotiating various options and capabilities, and communicating with and controlling the various codecs used. 7 Service Requirements 7.1 S

49、ystem VT-01. Backward Compatibility: Packet-switched VT system in each successive version of the network should be interoperable with the packet-switched VT in the previous version (e.g., between HRPD rev. 0 and rev. A). VT-02. Interworking. The VT system should be interoperable with international standardized video conferencing systems (e.g., ITU-T H.324M 2/ H.323 1, IETF RFC 3261 SIP 8). It should be able to interconnect with 3GPP circuit-switched VT. It should be able to exchange controlling information with peers in format complying with