NGN环境下多媒体会议系统几项关键技术的研究
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摘要
下一代网络(Next Generation Network,NGN)是电信网络的未来发展方向。NGN的提出和发展为多媒体交互技术的成熟和演进提供良好的基础架构级别的支持。本文旨在充分理解和利用NGN为多媒体交互业务发展带来的契机,在NGN环境下的多媒体会议系统的几方面关键技术进行了较为深入的研究、以及与之对应的工程实践。
多媒体会议的出现已经有较长的时间,其基于企业级的应用已经比较成熟。本文结合较长时间的跟踪研究,运用数学手段相对深入的研究和分析了多媒体会议的形式,给出了ACPMP功能模型,并证明、归纳和总结出设计高性能、高并发支持的NGN环境下的多媒体会议系统设计实现的八条原则。在理论分析的基础之上,提出的“系统熟识率”概念,用于评价系统用户的关系构成和系统的应用导向。
本文分析了现有主流多媒体会议标准,总结出其发展趋势是以松耦合、多应用、高扩展性的SIP协议将成为主导。根据在NGN环境下提供大规模多媒体会议服务的具体需求,运用这些原则提出了可以采用SIP承载的GMIP协议。我们给出了GMIP协议应用中的实体定义、信令参考模型和媒体参考模型,规约了GMIP协议PDU基于XML的定义,并且简述了PDU的定义和生成方式,最后给出了GMIP协议栈的面向真实部署的实现。
本文集中的探讨了NGN环境下多媒体会议系统的高性能、实时媒体合成的模型和算法,在混音方案中,我们构造出了ASW、AEW两种新的输入加权混音算法,还对其进行面向一般的混音模型的优化。我们还给出了代数和分段加权算法,通过对该算法控制参数的分析,最后给出了两种实用方案的优化算法。对于视频合成,我们提出了一般性的模型,并且给出DCT域的视频合成算法,结合对应的运动欠量合成算法,我们给出一般性的视频合成和转码流程。在NAT穿透方面,通过分析和研究现有的MIDCOM、STUN、TURN、ICE等多种NAT穿透方案,我们明确提出对于SIP/GMIP信令统一采用中转形式,而对于媒体则尽量采用NAT穿透。我们给出与呼叫过程结合的优化的媒体NAT穿越方案,并且将其与现有方案进行比对,并且将其与GMIP整合构成整体架构的一个部分。
最后,本文简述了对目前多套已经成功商用的多媒体通信的反向工程结果,并对这些结果进行了分析,总结它们的优势和不足。结合我们对于NGN环境下的多媒体会议形式、协议、媒体处理和传输方面的研究结果,并将展开大规模多媒体会议服务运营作为系统实现的重要目标,给出了GMI系统的相关模型、方法论和架构,并且对于GMI系统的实现做了扼要的示例。最后将GMI系统与现有的多套成功商用的多媒体通信系统进行逐项对比,明确GMI系统的在NGN环境下的优势,同时也看到不足。
Next Generation Network (NGN) is the direction of development of the telecommunication network. The emergence and development of NGN provides solid supports in the infrastructure level for the maturity and evolution of multimedia interactivity technology.It is relatively long time for the emergence of multimedia conferencing. The enterprise level application of it is very mature. In this thesis, according to our research work within a long period, we analyzed the form of multimedia conferencing with mathematical tools, and we bring forward the ACPMP functional model. And we demonstrated, induced and summarized eight principles for design of high performance and high simultaneity multimedia conferencing system in NGN environment. Based on theoretical analysis, we defined "System Acquaintance ratio", which is used to evaluate the user relation composition and the application orientation of the whole system.We analyzed several main standard families related to multimedia conferencing in this thesis. We concluded that the trend is for SIP, which is loose coupled, multi-application supported, and highly scalable. To meet the requirements of providing large scale multimedia conferencing service in the NGN environment, we defined GMIP, which can be carried by SIP, based on the eight principles. We defined the entities, provided the reference model for signaling and media interactivity, and regulated the PDLJ definition based on XML Schema, and provided the definition and generation method for those PDUs. Afterward we described how we implemented the GMIP stack for the real deployment.In this thesis, we concentratively discussed the high performance media mixing or combination algorithm. For audio mixing, we designed two new input-weighted algorithms, namely ASW and AEW, and then bring out the optimized audio mixing model for them. We also developed algebra sum segmented weighted algorithm, and after the analysis for the parameters, we proposed two algorithms for practical applications. With in the field of video combination, we give out the general model for video combination and transcoding, and then we developed two DCT domain resampling algorithms and the corresponding motion vector composition algorithms, then we proposed the generalized video combination and transcoding process. As for NAT traversal, we analyzed and studied many solutions, such as MIDCOM, STUN, TURN and ICE. We proposed that all the SIP/GMIP Signalling must be proxied or relayed by server-side entities, and media stream should do the NAT traversal work. We design an optimized NAT traversal solution for media stream combined with the signaling process. Then we compared our solution to other solutions, and made it a part of the whole architecture of GMIP.We do reverse engineering work to some famous multimedia communication system in real operation, and summarized the advantages and disadvantages of them. Using the results of the conferencing form, protocol, media processing and transportation, taking the aim which is to providing the large scale multimedia conferencing service, we bring out the models, methodology and architecture of GMI system, and we provided some samples for the implementation of GMI system. In the end, we compared GMI system to some successfully operated multimedia
communication systems, so as to find out the advantages of GMI system, and to see the shortcomings in the mean time.
多媒体会议的出现已经有较长的时间,其基于企业级的应用已经比较成熟。本文结合较长时间的跟踪研究,运用数学手段相对深入的研究和分析了多媒体会议的形式,给出了ACPMP功能模型,并证明、归纳和总结出设计高性能、高并发支持的NGN环境下的多媒体会议系统设计实现的八条原则。在理论分析的基础之上,提出的“系统熟识率”概念,用于评价系统用户的关系构成和系统的应用导向。
本文分析了现有主流多媒体会议标准,总结出其发展趋势是以松耦合、多应用、高扩展性的SIP协议将成为主导。根据在NGN环境下提供大规模多媒体会议服务的具体需求,运用这些原则提出了可以采用SIP承载的GMIP协议。我们给出了GMIP协议应用中的实体定义、信令参考模型和媒体参考模型,规约了GMIP协议PDU基于XML的定义,并且简述了PDU的定义和生成方式,最后给出了GMIP协议栈的面向真实部署的实现。
本文集中的探讨了NGN环境下多媒体会议系统的高性能、实时媒体合成的模型和算法,在混音方案中,我们构造出了ASW、AEW两种新的输入加权混音算法,还对其进行面向一般的混音模型的优化。我们还给出了代数和分段加权算法,通过对该算法控制参数的分析,最后给出了两种实用方案的优化算法。对于视频合成,我们提出了一般性的模型,并且给出DCT域的视频合成算法,结合对应的运动欠量合成算法,我们给出一般性的视频合成和转码流程。在NAT穿透方面,通过分析和研究现有的MIDCOM、STUN、TURN、ICE等多种NAT穿透方案,我们明确提出对于SIP/GMIP信令统一采用中转形式,而对于媒体则尽量采用NAT穿透。我们给出与呼叫过程结合的优化的媒体NAT穿越方案,并且将其与现有方案进行比对,并且将其与GMIP整合构成整体架构的一个部分。
最后,本文简述了对目前多套已经成功商用的多媒体通信的反向工程结果,并对这些结果进行了分析,总结它们的优势和不足。结合我们对于NGN环境下的多媒体会议形式、协议、媒体处理和传输方面的研究结果,并将展开大规模多媒体会议服务运营作为系统实现的重要目标,给出了GMI系统的相关模型、方法论和架构,并且对于GMI系统的实现做了扼要的示例。最后将GMI系统与现有的多套成功商用的多媒体通信系统进行逐项对比,明确GMI系统的在NGN环境下的优势,同时也看到不足。
Next Generation Network (NGN) is the direction of development of the telecommunication network. The emergence and development of NGN provides solid supports in the infrastructure level for the maturity and evolution of multimedia interactivity technology.It is relatively long time for the emergence of multimedia conferencing. The enterprise level application of it is very mature. In this thesis, according to our research work within a long period, we analyzed the form of multimedia conferencing with mathematical tools, and we bring forward the ACPMP functional model. And we demonstrated, induced and summarized eight principles for design of high performance and high simultaneity multimedia conferencing system in NGN environment. Based on theoretical analysis, we defined "System Acquaintance ratio", which is used to evaluate the user relation composition and the application orientation of the whole system.We analyzed several main standard families related to multimedia conferencing in this thesis. We concluded that the trend is for SIP, which is loose coupled, multi-application supported, and highly scalable. To meet the requirements of providing large scale multimedia conferencing service in the NGN environment, we defined GMIP, which can be carried by SIP, based on the eight principles. We defined the entities, provided the reference model for signaling and media interactivity, and regulated the PDLJ definition based on XML Schema, and provided the definition and generation method for those PDUs. Afterward we described how we implemented the GMIP stack for the real deployment.In this thesis, we concentratively discussed the high performance media mixing or combination algorithm. For audio mixing, we designed two new input-weighted algorithms, namely ASW and AEW, and then bring out the optimized audio mixing model for them. We also developed algebra sum segmented weighted algorithm, and after the analysis for the parameters, we proposed two algorithms for practical applications. With in the field of video combination, we give out the general model for video combination and transcoding, and then we developed two DCT domain resampling algorithms and the corresponding motion vector composition algorithms, then we proposed the generalized video combination and transcoding process. As for NAT traversal, we analyzed and studied many solutions, such as MIDCOM, STUN, TURN and ICE. We proposed that all the SIP/GMIP Signalling must be proxied or relayed by server-side entities, and media stream should do the NAT traversal work. We design an optimized NAT traversal solution for media stream combined with the signaling process. Then we compared our solution to other solutions, and made it a part of the whole architecture of GMIP.We do reverse engineering work to some famous multimedia communication system in real operation, and summarized the advantages and disadvantages of them. Using the results of the conferencing form, protocol, media processing and transportation, taking the aim which is to providing the large scale multimedia conferencing service, we bring out the models, methodology and architecture of GMI system, and we provided some samples for the implementation of GMI system. In the end, we compared GMI system to some successfully operated multimedia
communication systems, so as to find out the advantages of GMI system, and to see the shortcomings in the mean time.
引文
[1] IETF RFC 2543, "SIP: Session Initiation Protocol (obsolete)", March, 1999
[2] IETF RFC 3261, "SIP: Session Initiation Protocol", June, 2002
[3] IETF RFC 2327, "Session Description Protocol (SDP)", April, 1998
[4] IETF RFC 3264, "An Offer/Answer Model with the Session Description Protocol (SDP)", June, 2002
[5] IETF RFC 3266, "Support of IPv6 in SDP", June, 2002
[6] IETF RFC 3388, "Grouping of Media Lines in the Session Description Protocol (SDP)", Dec, 2002
[7] IETF RFC 3407, "Session Description Protocol (SDP) Simple Capability Declaration", Oct, 2002
[8] IETF RFC 3556, "Session Description Protocol (SDP) Bandwidth Modifiers for RTCP Bandwidth", July, 2003
[9] IETF RFC 3605, "Real Time Control Protocol (RTCP) attribute in Session Description Protocol (SDP)", Oct, 2003
[10] IETF RFC 3890, "A Transport Independent Bandwidth Modifier for the Session Description Protocol (SDP)", Sept, 2004
[11] IETF RFC 4091, "The Alternative Network Address Types (ANAT) Semantics for the Session Description Protocol (SDP) Grouping Framework ", June, 2005
[12] IETF RFC 1889, "RTP: Transport Protocol for Real-Time Applications (obsolete)", Jan, 1996
[13] IETF RFC 1890, "RTP Profile for Audio and Video Conferences with Minimal Control (obsolete)", Jan, 1996
[14] IETF RFC 3550, "RTP: Transport Protocol for Real-Time Applications", July, 2003
[15] IETF RFC 3551, "RTP Profile for Audio and Video Conferences with Minimal Control", July, 2003
[16] IETF RFC 2198, "RTP Payload for Redundant Audio Data", Sept, 1997
[17] IETF RFC 2733, "An RTP Payload Format for Generic Forward Error Correction", Dec, 1999
[18] IETF RFC 2793, "RTP Payload for Text Conversation", May, 2000
[19] IETF RFC 2833, "RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals", May, 2000
[20] IETF RFC 2959, "Real-Time Transport Protocol Management Information Base", Oct, 2000
[21] IETF RFC 3389, "RTP Payload for Comfort Noise (CN)", Sept, 2002
[22] IETF RFC 3611, "RTP Control Protocol Extended Reports (RTCP XR)", Nov, 2003
[23] IETF RFC 2032, "RTP Payload Format for H.261 Video Streams", Oct, 1996
[24] IETF RFC 2190, Zhu, C., "RTP Payload Format for H.263 Video Streams", Sept, 1997
[25] IETF RFC 2429, "RTP Payload Format for the 1998 Version of ITU-T Rec. H.263 Video (H.263+)", Oct, 1998
[26] IETF RFC 3984, "RTP payload Format for H.264 Video", Feb, 2004
[27] IETF RFC 3711, "The Secure Real-time Transport Protocol (SRTP)", March, 2004
[28] IETF RFC 2616, "Hypertext Transfer Protocol—HTTP/1.1", June, 1999
[29] IETF RFC 2617, "HTTP Authentication: Basic and Digest Access Authentication", June, 1999
[30] IETF RFC 3310, "HTTP Digest Authentication Using Authentication and Key Agreement (AKA)", Sept, 2002
[31] IETF RFC 1847, "Security Multiparts for MIME: Muitipart/Signed and Multipart/Encrypted", Oct, 1995
[32] IETF RFC 2045, "Multipurpose Internet Mail Extensions (MIME) Part One: Format of Internet Message Bodies", Mov, 1996
[33] IETF RFC 2046, "Multipurpose Internet Mail Extensions (MIME) Part Two: Media Types", Nov, 1996
[34] IETF RFC 2047, "MIME (Multipurpose Internet Mail Extensions) Part Three: Message Header Extensions for Non-ASCⅡ Text", Nov, 1996
[35] IETF RFC 2048, "Multipurpose Internet Mail Extensions (MIME) Part Four: Registration Procedures", Nov, 1996
[36] IETF RFC 2633, "S/MIME Version 3 Message Specification", June, 1999
[37] IETF RFC 3204, "MIME media types for ISUP and QSIG Objects", Dec, 2001
[38] IETF RFC 3420, "Internet Media Type message/sipfrag", Nov, 2002
[39] IETF RFC 3555, "MIME Type Registration of RTP Payload Formats", July, 2003
[40] IETF RFC 2976, "The SIP INFO Method", Oct, 2000
[41] IETF RFC 2848, "The PINT Service Protocol: Extensions to SIP and SDP for IP Access to Telephone Call Services", June, 2000
[42] IETF RFC 3050, "Common Gateway Interface for SIP", Jan, 2001
[43] IETF RFC 3262, "Reliability of Provisional Responses in Session Initiation Protocol (SIP)", June, 2002
[44] IETF RFC 3263, "Session Initiation Protocol (SIP): Locating SlP Servers", June, 2002
[45] IETF RFC 3265, "Session Initiation Protocol (SIP)-Specific Event Notification", June, 2002
[46] IETF RFC 3311, "The Session Initiation Protocol (SIP) UPDATE Method", Sept, 2002
[47] IETF RFC 3312, "Integration of Resource Management and Session Initiation Protocol (SIP)", Oct, 2002
[48] IETF RFC 3313, "Private Session Initiation Protocol (SIP) Extensions for Media Authorization", Jan, 2003
[49] IETF RFC 3319, "Dynamic Host Configuration Protocol (DHCPv6) Options for Session Initiation Protocol (SIP) Servers", July, 2003
[50] IETF RFC 3323, "A Privacy Mechanism for the Session Initiation Protocol (SIP)", Nov, 2002
[51] IETF RFC 3324, "Short Term Requirements for Network Asserted Identity", Nov, 2002
[52] IETF RFC 3325, "Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks", Nov, 2002
[53] IETF RFC 3326, "The Reason Header Field for the Session Initiation Protocol (SIP) ", Dec, 2002
[54] IETF RFC 3327, "Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts", Dec, 2002
[55] IETF RFC 3329, "Security Mechanism Agreement for the Session Initiation Protocol (SIP)", Jan, 2003
[56] IETF RFC 3361, "Dynamic Host Configuration Protocol (DHCP-for-IPv4) Option for Session Initiation Protocol (SIP) Servers", Aug, 2002
[57] IETF RFC 3372, "Session Initiation Protocol for Telephones (SIP-T): Context and Architectures", Sept, 2002
[58] IETF RFC 3398, "Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping", Dec, 2002
[59] IETF RFC 3428, "Session Initiation Protocol (SIP) Extension for Instant Messaging", Dec, 2002
[60] IETF RFC 3455, "Private Header (P-Header) Extensions to the Session Initiation Protocol (SIP) for the 3rd-Generation Partnership Project (3GPP)", Jan, 2003
[61] IETF RFC 3515, "The Session Initiation Protocol (SIP) Refer Method", April, 2003
[62] IETF RFC 3608, "Session Initiation Protocol (SIP) Extension Header Field for Service Route Discovery During Registration", Oct, 2003
[63] IETF RFC 3680, "A Session Initiation Protocol (SIP) Event Package for Registrations", March, 2004
[64] IETF RFC 3840, "Indicating User Agent Capabilities in the Session Initiation Protocol (SIP) ", Aug, 2004
[65] IETF RFC 3841, "Caller Preferences for the Session Initiation Protocol (SIP)", Aug, 2004
[66] IETF RFC 3842, "A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)", Aug, 2004
[67] IETF RFC 3856, "A Presence Event Package for the Session Initiation Protocol (SIP)", Aug, 2004
[68] IETF RFC 3857, "A Watcher Information Event Template-Package for the Session Initiation Protocol (SIP)", Aug, 2004
[69] IETF RFC 3858, "An Extensible Markup Language (XML) Based Format for Watcher Information", Aug, 2004
[70] IETF RFC 3863, "Presence Information Data Format (PIDF)", Aug, 2004
[71] IETF RFC 3891, "The Session Initiation Protocol (SIP) "Replaces" Header ", Sept, 2004
[72] IETF RFC 3903, "Session Initiation Protocol (SIP) Extension for Event State Publication", Oct, 2004
[73] IETF RFC 3959, "The Early Session Disposition Type for the Session Initiation Protocol (SIP)", Dec, 2004
[74] IETF RFC 3960, "Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)", Dec, 2004
[75] IETF RFC 4028, "Session Timers in the Session Initiation Protocol (SIP)", April, 2005
[76] IETF RFC 3087, "Control of Service Context using SIP Request-URI", April, 2001
[77] IETF RFC 3351, "User Requirements for the Session Initiation Protocol (SIP) in Support of Deaf, Hard of Hearing and Speech-impaired Individuals", Aug, 2002
[78] IETF RFC 3603, "Private Session Initiation Protocol (SIP) Proxy-to-Proxy Extensions for Supporting the PacketCable Distributed Call Signaling Architecture", Oct, 2003
[79] IETF RFC 3702, "Authentication, Authorization, and Accounting Requirements for the Session Initiation Protocol (SIP)", Feb, 2004
[80] IETF RFC 3824, "Using E.164 numbers with the Session Initiation Protocol (SIP)", June, 2004
[81] IETF RFC 3911, "The Session Initiation Protocol (SIP) "Join" Header", Oct, 2004
[82] IETF RFC 3968, "The Internet Assigned Number Authority (IANA) Header Field Parameter Registry for the Session Initiation Protocol (SIP)", Dec, 2004
[83] IETF RFC 3969, "The Internet Assigned Number Authority (IANA) Uniform Resource Identifier (URI) Parameter Registry for the Session Initiation Protocol (SIP)", Dec, 2004
[84] IETF RFC 3976, "Interworking SIP and Intelligent Network (IN) Applications", Jan, 2005
[85] IETF RFC 4117, "Transcoding Services Invocation in the Session Initiation Protocol (SIP) Using Third Party Call Control (3pcc)", June, 2005
[86] IETF RFC 4123, "Session Initiation Protocol (SIP)-H.323 Interworking Requirements", July, 2005
[87] IETF RFC 3219, "Telephony Routing over IP (TRIP)", Jan, 2002
[88] IETF RFC 3320, "Signaling Compression (SigComp)", Jan, 2003
[89] IETF RFC 3321, "Signaling Compression (SigComp)-Extended Operations", Jan, 2003
[90] IETF RFC 3322, "Signaling Compression (SigComp) Requirements & Assumptions", Jan, 2003
[91] IETF RFC 3486, "Compressing the Session Initiation Protocol (SIP)", Feb, 2003
[92] IETF RFC 3485, "The Session Initiation Protocol (SIP) and Session Description Protocol (SDP) Static Dictionary for Signaling Compression (SigComp)", Feb, 2003
[93] IETF RFC 3725, "Best Current Practices for Third Party Call Control (3pcc) in the Session Initiation Protocol (SIP)", April, 2004
[94] IETF RFC 3764, "enumservice registration for Session Initiation Protocol (SIP) Addresses-of-Record", April, 2004
[95] IETF RFC 4077, "A Negative Acknowledgement Mechanism for Signaling Compression", May, 2005
[96] IETF RFC 4083, "Input 3rd-Generation Partnership Project (3GPP) Release 5 Requirements on the Session Initiation Protocol (SIP)", May, 2005
[97] IETF RFC 4092, "Usage of the Session Description Protocol (SDP) Alternative Network Address Types (ANAT) Semantics in the Session Initiation Protocol (SIP)", June, 2005
[98] IETF RFC 3235, "Network Address Translator (NAT)-Friendly Application Design Guidelines", Jan, 2002
[99] IETF RFC 3303, "Middlebox communication architecture and framework", Aug, 2002
[100] IETF RFC 3304, "Middlebox Communications (midcom) Protocol Requirements", Aug, 2002
[101] IETF RFC 3489, "STUN-Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)", March, 2003
[102] IETF RFC 3989, "Middlebox Communications (MIDCOM) Protocol Semantics", Feb, 2005
[103] IETF RFC 4097, "Middlebox Communications (MIDCOM) Protocol Evaluation", June, 2005
[104] IETF RFC 3880, "Call Processing Language (CPL): A Language for User Control of Intemet Telephony Services", Oct, 2004
[105] IETF RFC 3851, "Internet Low Bit Rate Codec (iLBC)", Dec, 2004
[106] IETF RFC 1631, "The IP Network Address Translator (NAT)", May, 1994
[107] IETF RFC 1918, "Address Allocation for Private Internets", Feb, 1996
[108] IETF RFC 3424, "IAB Considerations for UNilateral Self-Address Fixing (UNSAF) Across Network Address Translation", Nov, 2002
[109] ITU-T Recommendation G.711, "Pulse code modulation (PCM) of voice frequencies", Nov, 1988
[110] ITU-T Recommendation G.722, "7 kHz audio-coding within 64 kbit/s", Nov, 1988
[111] ITU-T Recommendation G.722.1 Annex A, "Packet format, capability identifiers and capability parameters", Feb, 2000
[112] ITU-T Recommendation G.722.1, "Coding at 24 and 32 kbit/s for hands-free operation in systems with low frame loss", Sept, 1999
[113] ITU-T Recommendation G.722.2, "Wideband coding of speech at around 16 kbit/s using adaptive multi-rate wideband (AMR-WB)", Jan, 2002
[114] ITU-T Recommendation G.722.2 Annex A, "Comfort noise aspects", Jan, 2002
[115] ITU-T Recommendation G.722.2 Annex B, "Source controlled rate operation", Jan, 2002
[116] ITU-T Recommendation G.722.2 Annex C, "Fixed-point C-code", Jan, 2002
[117] ITU-T Recommendation G.722.2 Annex E, "Frame structure", Jan, 2002
[118] ITU-T Recommendation G.723.1 "Speech coders: Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s", March, 1996
[119] ITU-T Recommendation G.728, "Coding of speech at 16 kbit/s using low-delay code excited linear prediction", Sept, 1992
[120] ITU-T Recommendation G.729, "Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP)", March, 1996
[121] ITU-T Recommendation G.729 Annex A, "Reduced complexity 8 kbit/s CS-ACELP speech codec", Nov, 1996
[122] ITU-T Recommendation H. 100, "Visual telephone systems", Nov, 1988
[123] ITU-T Recommendation H. 110, "Hypothetical reference connections for videoconferencing using primary digital group transmission", Nov, 1988
[124] ITU-T Recommendation H.120, "Codecs for videoconferencing using primary digital group transmission", Mar, 1993
[125] ITU-T Recommendation H.130, "Frame structures for use in the international interconnection of digital codecs for videoconferencing or visual telephony", Nov, 1988
[126] ITU-T Recommendation H.140, "multipoint international videoconference system", Nov, 1988
[127] ITU-T Recommendation H.225.0, "Call signalling protocols and media stream packetization for packet-based multimedia communication systems", Nov, 2000
[128] ITU-T Recommendation H.231, "Multipoint control units for audiovisual systems using digital channels up to 1920 kbit/s", July, 1997
[129] ITU-T Recommendation H.235, "Security and encryption for H-Series (H.323 and other H.245-based) multimedia terminals", Nov, 2000
[130] ITU-T Recommendation H.241, "Extended video procedures and control signals for H.300 SERIES TERMINALS", July, 2003
[131] ITU-T Recommendation H.243, "Procedures for establishing communication between three or more audiovisual terminals using digital channels up to 1920 kbit/s", Feb, 2000
[132] ITU-T Recommendation H.245, "Control Protocol for multimedia communication", July, 2001
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[249] ET, http://et.263.net
[250] GAIM Project, http://gaim.sourceforge.net/
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[252] Google Talk, http://www.google.com/talk/
[253] Jabber, Jabber Software Foundation, http://www.iabber.org
[254] JSLEE and the JAIN Initiative, http://java.sun.com/products/jain/
[255] Microsott Live Communications Server, http://office.microsoft.com/home/office.aspx?assetid=FX010908711033
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[257] OpenWengo, http://www.openwengo.com
[258] QQ, http://www.qq.com
[259] ReSIProcate Project, http://www.sipfoundrv.org/reSIProcate/
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[261] SIPImp Project, http://sourceforge.net/projects/sipimp/
[262] Skype, Skype Technologies S. A., http://www.skype.com
[263] Teltel, http://www.teltel.com
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[265] The Parlay Group, http://www.parlay.org
[266] VOCAL Project, http://www.vovida.org
[267] vSkype, Santa Cruz Networks, http://www.vskype.com
[268] Webex, http://www.webex.com
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[2] IETF RFC 3261, "SIP: Session Initiation Protocol", June, 2002
[3] IETF RFC 2327, "Session Description Protocol (SDP)", April, 1998
[4] IETF RFC 3264, "An Offer/Answer Model with the Session Description Protocol (SDP)", June, 2002
[5] IETF RFC 3266, "Support of IPv6 in SDP", June, 2002
[6] IETF RFC 3388, "Grouping of Media Lines in the Session Description Protocol (SDP)", Dec, 2002
[7] IETF RFC 3407, "Session Description Protocol (SDP) Simple Capability Declaration", Oct, 2002
[8] IETF RFC 3556, "Session Description Protocol (SDP) Bandwidth Modifiers for RTCP Bandwidth", July, 2003
[9] IETF RFC 3605, "Real Time Control Protocol (RTCP) attribute in Session Description Protocol (SDP)", Oct, 2003
[10] IETF RFC 3890, "A Transport Independent Bandwidth Modifier for the Session Description Protocol (SDP)", Sept, 2004
[11] IETF RFC 4091, "The Alternative Network Address Types (ANAT) Semantics for the Session Description Protocol (SDP) Grouping Framework ", June, 2005
[12] IETF RFC 1889, "RTP: Transport Protocol for Real-Time Applications (obsolete)", Jan, 1996
[13] IETF RFC 1890, "RTP Profile for Audio and Video Conferences with Minimal Control (obsolete)", Jan, 1996
[14] IETF RFC 3550, "RTP: Transport Protocol for Real-Time Applications", July, 2003
[15] IETF RFC 3551, "RTP Profile for Audio and Video Conferences with Minimal Control", July, 2003
[16] IETF RFC 2198, "RTP Payload for Redundant Audio Data", Sept, 1997
[17] IETF RFC 2733, "An RTP Payload Format for Generic Forward Error Correction", Dec, 1999
[18] IETF RFC 2793, "RTP Payload for Text Conversation", May, 2000
[19] IETF RFC 2833, "RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals", May, 2000
[20] IETF RFC 2959, "Real-Time Transport Protocol Management Information Base", Oct, 2000
[21] IETF RFC 3389, "RTP Payload for Comfort Noise (CN)", Sept, 2002
[22] IETF RFC 3611, "RTP Control Protocol Extended Reports (RTCP XR)", Nov, 2003
[23] IETF RFC 2032, "RTP Payload Format for H.261 Video Streams", Oct, 1996
[24] IETF RFC 2190, Zhu, C., "RTP Payload Format for H.263 Video Streams", Sept, 1997
[25] IETF RFC 2429, "RTP Payload Format for the 1998 Version of ITU-T Rec. H.263 Video (H.263+)", Oct, 1998
[26] IETF RFC 3984, "RTP payload Format for H.264 Video", Feb, 2004
[27] IETF RFC 3711, "The Secure Real-time Transport Protocol (SRTP)", March, 2004
[28] IETF RFC 2616, "Hypertext Transfer Protocol—HTTP/1.1", June, 1999
[29] IETF RFC 2617, "HTTP Authentication: Basic and Digest Access Authentication", June, 1999
[30] IETF RFC 3310, "HTTP Digest Authentication Using Authentication and Key Agreement (AKA)", Sept, 2002
[31] IETF RFC 1847, "Security Multiparts for MIME: Muitipart/Signed and Multipart/Encrypted", Oct, 1995
[32] IETF RFC 2045, "Multipurpose Internet Mail Extensions (MIME) Part One: Format of Internet Message Bodies", Mov, 1996
[33] IETF RFC 2046, "Multipurpose Internet Mail Extensions (MIME) Part Two: Media Types", Nov, 1996
[34] IETF RFC 2047, "MIME (Multipurpose Internet Mail Extensions) Part Three: Message Header Extensions for Non-ASCⅡ Text", Nov, 1996
[35] IETF RFC 2048, "Multipurpose Internet Mail Extensions (MIME) Part Four: Registration Procedures", Nov, 1996
[36] IETF RFC 2633, "S/MIME Version 3 Message Specification", June, 1999
[37] IETF RFC 3204, "MIME media types for ISUP and QSIG Objects", Dec, 2001
[38] IETF RFC 3420, "Internet Media Type message/sipfrag", Nov, 2002
[39] IETF RFC 3555, "MIME Type Registration of RTP Payload Formats", July, 2003
[40] IETF RFC 2976, "The SIP INFO Method", Oct, 2000
[41] IETF RFC 2848, "The PINT Service Protocol: Extensions to SIP and SDP for IP Access to Telephone Call Services", June, 2000
[42] IETF RFC 3050, "Common Gateway Interface for SIP", Jan, 2001
[43] IETF RFC 3262, "Reliability of Provisional Responses in Session Initiation Protocol (SIP)", June, 2002
[44] IETF RFC 3263, "Session Initiation Protocol (SIP): Locating SlP Servers", June, 2002
[45] IETF RFC 3265, "Session Initiation Protocol (SIP)-Specific Event Notification", June, 2002
[46] IETF RFC 3311, "The Session Initiation Protocol (SIP) UPDATE Method", Sept, 2002
[47] IETF RFC 3312, "Integration of Resource Management and Session Initiation Protocol (SIP)", Oct, 2002
[48] IETF RFC 3313, "Private Session Initiation Protocol (SIP) Extensions for Media Authorization", Jan, 2003
[49] IETF RFC 3319, "Dynamic Host Configuration Protocol (DHCPv6) Options for Session Initiation Protocol (SIP) Servers", July, 2003
[50] IETF RFC 3323, "A Privacy Mechanism for the Session Initiation Protocol (SIP)", Nov, 2002
[51] IETF RFC 3324, "Short Term Requirements for Network Asserted Identity", Nov, 2002
[52] IETF RFC 3325, "Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks", Nov, 2002
[53] IETF RFC 3326, "The Reason Header Field for the Session Initiation Protocol (SIP) ", Dec, 2002
[54] IETF RFC 3327, "Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts", Dec, 2002
[55] IETF RFC 3329, "Security Mechanism Agreement for the Session Initiation Protocol (SIP)", Jan, 2003
[56] IETF RFC 3361, "Dynamic Host Configuration Protocol (DHCP-for-IPv4) Option for Session Initiation Protocol (SIP) Servers", Aug, 2002
[57] IETF RFC 3372, "Session Initiation Protocol for Telephones (SIP-T): Context and Architectures", Sept, 2002
[58] IETF RFC 3398, "Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping", Dec, 2002
[59] IETF RFC 3428, "Session Initiation Protocol (SIP) Extension for Instant Messaging", Dec, 2002
[60] IETF RFC 3455, "Private Header (P-Header) Extensions to the Session Initiation Protocol (SIP) for the 3rd-Generation Partnership Project (3GPP)", Jan, 2003
[61] IETF RFC 3515, "The Session Initiation Protocol (SIP) Refer Method", April, 2003
[62] IETF RFC 3608, "Session Initiation Protocol (SIP) Extension Header Field for Service Route Discovery During Registration", Oct, 2003
[63] IETF RFC 3680, "A Session Initiation Protocol (SIP) Event Package for Registrations", March, 2004
[64] IETF RFC 3840, "Indicating User Agent Capabilities in the Session Initiation Protocol (SIP) ", Aug, 2004
[65] IETF RFC 3841, "Caller Preferences for the Session Initiation Protocol (SIP)", Aug, 2004
[66] IETF RFC 3842, "A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)", Aug, 2004
[67] IETF RFC 3856, "A Presence Event Package for the Session Initiation Protocol (SIP)", Aug, 2004
[68] IETF RFC 3857, "A Watcher Information Event Template-Package for the Session Initiation Protocol (SIP)", Aug, 2004
[69] IETF RFC 3858, "An Extensible Markup Language (XML) Based Format for Watcher Information", Aug, 2004
[70] IETF RFC 3863, "Presence Information Data Format (PIDF)", Aug, 2004
[71] IETF RFC 3891, "The Session Initiation Protocol (SIP) "Replaces" Header ", Sept, 2004
[72] IETF RFC 3903, "Session Initiation Protocol (SIP) Extension for Event State Publication", Oct, 2004
[73] IETF RFC 3959, "The Early Session Disposition Type for the Session Initiation Protocol (SIP)", Dec, 2004
[74] IETF RFC 3960, "Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)", Dec, 2004
[75] IETF RFC 4028, "Session Timers in the Session Initiation Protocol (SIP)", April, 2005
[76] IETF RFC 3087, "Control of Service Context using SIP Request-URI", April, 2001
[77] IETF RFC 3351, "User Requirements for the Session Initiation Protocol (SIP) in Support of Deaf, Hard of Hearing and Speech-impaired Individuals", Aug, 2002
[78] IETF RFC 3603, "Private Session Initiation Protocol (SIP) Proxy-to-Proxy Extensions for Supporting the PacketCable Distributed Call Signaling Architecture", Oct, 2003
[79] IETF RFC 3702, "Authentication, Authorization, and Accounting Requirements for the Session Initiation Protocol (SIP)", Feb, 2004
[80] IETF RFC 3824, "Using E.164 numbers with the Session Initiation Protocol (SIP)", June, 2004
[81] IETF RFC 3911, "The Session Initiation Protocol (SIP) "Join" Header", Oct, 2004
[82] IETF RFC 3968, "The Internet Assigned Number Authority (IANA) Header Field Parameter Registry for the Session Initiation Protocol (SIP)", Dec, 2004
[83] IETF RFC 3969, "The Internet Assigned Number Authority (IANA) Uniform Resource Identifier (URI) Parameter Registry for the Session Initiation Protocol (SIP)", Dec, 2004
[84] IETF RFC 3976, "Interworking SIP and Intelligent Network (IN) Applications", Jan, 2005
[85] IETF RFC 4117, "Transcoding Services Invocation in the Session Initiation Protocol (SIP) Using Third Party Call Control (3pcc)", June, 2005
[86] IETF RFC 4123, "Session Initiation Protocol (SIP)-H.323 Interworking Requirements", July, 2005
[87] IETF RFC 3219, "Telephony Routing over IP (TRIP)", Jan, 2002
[88] IETF RFC 3320, "Signaling Compression (SigComp)", Jan, 2003
[89] IETF RFC 3321, "Signaling Compression (SigComp)-Extended Operations", Jan, 2003
[90] IETF RFC 3322, "Signaling Compression (SigComp) Requirements & Assumptions", Jan, 2003
[91] IETF RFC 3486, "Compressing the Session Initiation Protocol (SIP)", Feb, 2003
[92] IETF RFC 3485, "The Session Initiation Protocol (SIP) and Session Description Protocol (SDP) Static Dictionary for Signaling Compression (SigComp)", Feb, 2003
[93] IETF RFC 3725, "Best Current Practices for Third Party Call Control (3pcc) in the Session Initiation Protocol (SIP)", April, 2004
[94] IETF RFC 3764, "enumservice registration for Session Initiation Protocol (SIP) Addresses-of-Record", April, 2004
[95] IETF RFC 4077, "A Negative Acknowledgement Mechanism for Signaling Compression", May, 2005
[96] IETF RFC 4083, "Input 3rd-Generation Partnership Project (3GPP) Release 5 Requirements on the Session Initiation Protocol (SIP)", May, 2005
[97] IETF RFC 4092, "Usage of the Session Description Protocol (SDP) Alternative Network Address Types (ANAT) Semantics in the Session Initiation Protocol (SIP)", June, 2005
[98] IETF RFC 3235, "Network Address Translator (NAT)-Friendly Application Design Guidelines", Jan, 2002
[99] IETF RFC 3303, "Middlebox communication architecture and framework", Aug, 2002
[100] IETF RFC 3304, "Middlebox Communications (midcom) Protocol Requirements", Aug, 2002
[101] IETF RFC 3489, "STUN-Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)", March, 2003
[102] IETF RFC 3989, "Middlebox Communications (MIDCOM) Protocol Semantics", Feb, 2005
[103] IETF RFC 4097, "Middlebox Communications (MIDCOM) Protocol Evaluation", June, 2005
[104] IETF RFC 3880, "Call Processing Language (CPL): A Language for User Control of Intemet Telephony Services", Oct, 2004
[105] IETF RFC 3851, "Internet Low Bit Rate Codec (iLBC)", Dec, 2004
[106] IETF RFC 1631, "The IP Network Address Translator (NAT)", May, 1994
[107] IETF RFC 1918, "Address Allocation for Private Internets", Feb, 1996
[108] IETF RFC 3424, "IAB Considerations for UNilateral Self-Address Fixing (UNSAF) Across Network Address Translation", Nov, 2002
[109] ITU-T Recommendation G.711, "Pulse code modulation (PCM) of voice frequencies", Nov, 1988
[110] ITU-T Recommendation G.722, "7 kHz audio-coding within 64 kbit/s", Nov, 1988
[111] ITU-T Recommendation G.722.1 Annex A, "Packet format, capability identifiers and capability parameters", Feb, 2000
[112] ITU-T Recommendation G.722.1, "Coding at 24 and 32 kbit/s for hands-free operation in systems with low frame loss", Sept, 1999
[113] ITU-T Recommendation G.722.2, "Wideband coding of speech at around 16 kbit/s using adaptive multi-rate wideband (AMR-WB)", Jan, 2002
[114] ITU-T Recommendation G.722.2 Annex A, "Comfort noise aspects", Jan, 2002
[115] ITU-T Recommendation G.722.2 Annex B, "Source controlled rate operation", Jan, 2002
[116] ITU-T Recommendation G.722.2 Annex C, "Fixed-point C-code", Jan, 2002
[117] ITU-T Recommendation G.722.2 Annex E, "Frame structure", Jan, 2002
[118] ITU-T Recommendation G.723.1 "Speech coders: Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s", March, 1996
[119] ITU-T Recommendation G.728, "Coding of speech at 16 kbit/s using low-delay code excited linear prediction", Sept, 1992
[120] ITU-T Recommendation G.729, "Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP)", March, 1996
[121] ITU-T Recommendation G.729 Annex A, "Reduced complexity 8 kbit/s CS-ACELP speech codec", Nov, 1996
[122] ITU-T Recommendation H. 100, "Visual telephone systems", Nov, 1988
[123] ITU-T Recommendation H. 110, "Hypothetical reference connections for videoconferencing using primary digital group transmission", Nov, 1988
[124] ITU-T Recommendation H.120, "Codecs for videoconferencing using primary digital group transmission", Mar, 1993
[125] ITU-T Recommendation H.130, "Frame structures for use in the international interconnection of digital codecs for videoconferencing or visual telephony", Nov, 1988
[126] ITU-T Recommendation H.140, "multipoint international videoconference system", Nov, 1988
[127] ITU-T Recommendation H.225.0, "Call signalling protocols and media stream packetization for packet-based multimedia communication systems", Nov, 2000
[128] ITU-T Recommendation H.231, "Multipoint control units for audiovisual systems using digital channels up to 1920 kbit/s", July, 1997
[129] ITU-T Recommendation H.235, "Security and encryption for H-Series (H.323 and other H.245-based) multimedia terminals", Nov, 2000
[130] ITU-T Recommendation H.241, "Extended video procedures and control signals for H.300 SERIES TERMINALS", July, 2003
[131] ITU-T Recommendation H.243, "Procedures for establishing communication between three or more audiovisual terminals using digital channels up to 1920 kbit/s", Feb, 2000
[132] ITU-T Recommendation H.245, "Control Protocol for multimedia communication", July, 2001
[133] ITU-T Recommendation H.246, "Interworking of H-Series multimedia terminals with H-Series multimedia terminals and voice/voiceband terminals on GSTN and ISDN", Feb, 1998
[134] ITU-T Recommendation H.246 Annex C, "ISDN User Part Function-H.225.0 Interworking", Feb, 2000
[135] ITU-T Recommendation H.246 Annex E.1, "General Inter-Working Function (IWF) between Mobile Application Part and H.225.0", Nov, 2000
[136] ITU-T Recommendation H.246 Annex E.2, "Inter-Working Function (IWF) between Ansi-41 (Americas) Mobile Application Part and H.225.0", Nov, 2000
[137] ITU-T Recommendation H.246 Annex F, "H.323-H.324 interworking", July, 2001
[138] ITU-T Recommendation H.248.1, "Gateway control protocol: Version 2", May, 2002
[139] ITU-T Recommendation H.248.19, "Gateway Control Protocol: Decomposed Muitipoint Control Unit, Audio, Video and Data Conferencing Packages", Feb, 2004
[140] ITU-T Recommendation H.261, "Video codec for audiovisual services at p x 64 kbit/s", March, 1993
[141] ITU-T Recommendation H.262, "Information technology-Generic coding of moving pictures and associated audio information: Video", Feb, 2000
[142] ITU-T Recommendation H.262 Amendment 1, "Amendment 1: Video elementary stream content description data", Nov, 2000
[143] ITU-T Recommendation H.263, "Video coding for low bit rate communication", May, 1996
[144] ITU-T Recommendation H.263, "Video coding for low bit rate communication", Feb, 1998
[145] ITU-T Recommendation H.263 Annex U, "Enhanced reference picture selection mode", Nov, 2000
[146] ITU-T Recommendation H.263 Annex V, "Data partitioned slice (DPS) ", Nov, 2000
[147] ITU-T Recommendation H.263 Annex W, "Additional supplemental enhancement information", Nov, 2000
[148] ITU-T Recommendation H.263 Appendix Ⅱ, "Recommended optional enhancement", June, 2001
[149] ITU-T Recommendation H.264, "Advanced video coding for generic audiovisual services", May, 2003
[150] ITU-T Recommendation H.320, "Narrow-band visual telephone systems and terminal equipment", May, 1999
[151] ITU-T Recommendation H.321, "Adaptation of H.320 visual telephone terminals to B-ISDN environments", Feb, 1998
[152] ITU-T Recommendation H.322, "Visual telephone systems and terminal equipment for local area networks which provide a guaranteed quality of service", Mar, 1996
[153] ITU-T Recommendation H.323, "Packet-Based Multimedia Communication System", Nov, 2000
[154] ITU-T Recommendation H. 350, "Directory services architecture for multimedia conferencing", Aug, 2003
[155] ITU-T Recommendation H. 350 attachment, "This attachment contains the LDIF files referred to in clauses 8.1 and 8.2 of ITU-T Rec. H. 350 (08-2003)", Aug, 2003
[156] ITU-T Recommendation H. 350. 1, Directory services architecture for H. 323", Aug, 2003
[157] ITU-T Recommendation H. 350. 1 attachment, "This attachment contains the LDIF files referred to in clause 7 of ITU-T Rec. H. 350. 1 (08-2003)", Aug, 2003
[158] ITU-T Recommendation H.350.4, "Directory services architecture for SIP", Aug, 2003
[159] ITU-T Recommendation H. 350. 4 attachment, "This attachment contains the LDIF files referred to in clause 7 of ITU-T Rec. H. 350. 4 (08-2003)", Aug, 2003
[160] ITU-T Recommendation Q. 931, "ISDN user-network interface layer 3 specification for basic call control", May, 1998
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[162] ITU-T Recommendation T. 120 Annex C, "Lightweight profiles for the T. 120 architecture", Feb, 1998
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[166] ITU-T Recommendation T. 123, "Network-specific data protocol stacks for multimedia conferencing", May, 1999
[167] ITU-T Recommendation T. 124, "Generic Conference Control", Feb, 1998
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[262] Skype, Skype Technologies S. A., http://www.skype.com
[263] Teltel, http://www.teltel.com
[264] The Chord Project, http://pdos.csail.mit.edu/chord/
[265] The Parlay Group, http://www.parlay.org
[266] VOCAL Project, http://www.vovida.org
[267] vSkype, Santa Cruz Networks, http://www.vskype.com
[268] Webex, http://www.webex.com
[269] Yahoo Messenger, http://messenger.yahoo.com
[270] 宝视通,中国联合通信有限公司,http://video.cnuninet.com
[271] 新视通,中国电信集团公司,http://www.chinatelecom.com.cn/20050311/00019231.html