SIPHello媒体栈中改善服务质量若干技术研究
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摘要
随着VoIP技术的不断发展,标准SIP终端的功能越来越丰富。作为一个自主研发的标准SIP终端,SIPHello的功能从简单的语音通话和即时消息等功能,发展到复杂的在线消息订阅和视频通话等功能,未来还将实现视频会议和IPTV等功能。在终端功能不断增加的同时,人们对VoIP系统的通信服务质量要求也越来越高。目前提高服务质量的研究有基于网络底层的改进,比如区分服务和综合服务;有基于网络传输的改进,比如Overlay网络以及应用层重新选路;还有基于应用程序的改进,比如自适应抖动缓冲区算法。这些改进方法都是基于目前网络带宽有限的前提下,在一定程度上减轻了网络拥堵时通话质量的下降程度。然而,在当前网络状况不断改善以及接入带宽不断提高的背景下,有必要从另外一个角度提出一些改进服务质量的方法。此外,随着网络技术和视频压缩技术的发展,用户在通信时越来越青睐于视频通信。目前SIPHello只支持简单的视频通话功能,且存在与其他终端互通性差的问题,因此需要我们做进一步的研究,解决兼容性和优化等问题。
本文的主要目标是通过增加音频的宽带采样、视频数据RTP封装标准化和视频采集的优化三方面的改进,提高SIPHello的服务质量和视频通信的能力。文中首先介绍了SIPHello协议栈和媒体栈中使用的相关协议和技术,包括信令协议、传输协议和媒体编解码技术。然后参考借鉴了VoIP系统中主流客户端引入宽带采样提高语音通话质量的成功经验,给出了提高媒体栈服务质量和视频互通性的改进方案。具体工作分为三个部分:将支持宽带采样的音频编解码Speex移植到SIPHello媒体栈中;针对SIPHello与其它SIP终端视频互通性差的问题,研究了H.263视频数据RTP封装的技术标准并改进了视频编解码器;研究基于DirectShow模式的视频采集方法,设计并优化了视频采集模块。本文详细介绍了设计方案的实现过程,并对SIPHello进行了音频服务质量和视频互通性两个方面的测试。
文章最后对测试数据进行了分析,结果显示:语音通话质量有了明显提高;视频通信方面,增加了H.263视频分片技术后,SIPHello与其他终端视频互通能力得到了较大的提高,实现了与大多数SIP终端视频互通的目的。
With the progress of VoIP technology, the function of standard SIP UA gets more and more variety. As a standard SIP UA under self development, SIPHello as a simple UA which support only Voice Communication and Instant Message, have developed to supporting Presence and Video Communication. In future, SIPHello plan to support more functions such as Video Conference and IPTV. In the mean time, people have more and more requirements on service quality. Common research on QoS include: improvements based on network layer such as Differentiated Services and Integrated Services; improvements based on network transporting such as Overlay and application layer rerouting; and improvements based on application such as jitter buffer algorithm. These methods are all based on a limited bandwidth Internet now, and improve quality of service when the network is congested. But we believe the Next Generation Network (NGN) will have unlimited bandwidth and guaranteed QoS, therefore we need to improve service quality based on NGN. In addition, as bandwidth resource getting more sufficient, the requirement of video communication of SIP UA is more and more important. SIPHello can only support simple video communication functions now, and it can not interwork with other SIP UA, so we need to research on compatibility and improvement on efficience.
This paper is based on this background. By improving on media stack of SIPHello, we enhance the quality of serviceof SIPHello. As in audio improvement, we refer to some common VoIP UA, make media stack to support high sample rate to improve voice quality. As in video communication, we research on H.263 and it's transporting protocol, plant a open source codec into SIPHello and make some improvements.
Firstly, this paper analyzes and researches some technologies in SIPHello in the round, including signal protocol, transporting protocol and codec. Then we research on some problems existed in SIPHello and propose some approaches to improve those problems. The whole work include three parts: make media stack to support 16KHz sample rate; based on some standards, modify the H.263 codec to support interworking with other SIP UA, and improve on video capture. This paper introduce all the realize procedure, and perform tests on voice quality improvement and video communication interworking.
The last part of this paper analyzes this approach and its' results on the test datas, the results show that: the voice quality enhanced significantly and SIPHello can interwork with most of SIP UA.
本文的主要目标是通过增加音频的宽带采样、视频数据RTP封装标准化和视频采集的优化三方面的改进,提高SIPHello的服务质量和视频通信的能力。文中首先介绍了SIPHello协议栈和媒体栈中使用的相关协议和技术,包括信令协议、传输协议和媒体编解码技术。然后参考借鉴了VoIP系统中主流客户端引入宽带采样提高语音通话质量的成功经验,给出了提高媒体栈服务质量和视频互通性的改进方案。具体工作分为三个部分:将支持宽带采样的音频编解码Speex移植到SIPHello媒体栈中;针对SIPHello与其它SIP终端视频互通性差的问题,研究了H.263视频数据RTP封装的技术标准并改进了视频编解码器;研究基于DirectShow模式的视频采集方法,设计并优化了视频采集模块。本文详细介绍了设计方案的实现过程,并对SIPHello进行了音频服务质量和视频互通性两个方面的测试。
文章最后对测试数据进行了分析,结果显示:语音通话质量有了明显提高;视频通信方面,增加了H.263视频分片技术后,SIPHello与其他终端视频互通能力得到了较大的提高,实现了与大多数SIP终端视频互通的目的。
With the progress of VoIP technology, the function of standard SIP UA gets more and more variety. As a standard SIP UA under self development, SIPHello as a simple UA which support only Voice Communication and Instant Message, have developed to supporting Presence and Video Communication. In future, SIPHello plan to support more functions such as Video Conference and IPTV. In the mean time, people have more and more requirements on service quality. Common research on QoS include: improvements based on network layer such as Differentiated Services and Integrated Services; improvements based on network transporting such as Overlay and application layer rerouting; and improvements based on application such as jitter buffer algorithm. These methods are all based on a limited bandwidth Internet now, and improve quality of service when the network is congested. But we believe the Next Generation Network (NGN) will have unlimited bandwidth and guaranteed QoS, therefore we need to improve service quality based on NGN. In addition, as bandwidth resource getting more sufficient, the requirement of video communication of SIP UA is more and more important. SIPHello can only support simple video communication functions now, and it can not interwork with other SIP UA, so we need to research on compatibility and improvement on efficience.
This paper is based on this background. By improving on media stack of SIPHello, we enhance the quality of serviceof SIPHello. As in audio improvement, we refer to some common VoIP UA, make media stack to support high sample rate to improve voice quality. As in video communication, we research on H.263 and it's transporting protocol, plant a open source codec into SIPHello and make some improvements.
Firstly, this paper analyzes and researches some technologies in SIPHello in the round, including signal protocol, transporting protocol and codec. Then we research on some problems existed in SIPHello and propose some approaches to improve those problems. The whole work include three parts: make media stack to support 16KHz sample rate; based on some standards, modify the H.263 codec to support interworking with other SIP UA, and improve on video capture. This paper introduce all the realize procedure, and perform tests on voice quality improvement and video communication interworking.
The last part of this paper analyzes this approach and its' results on the test datas, the results show that: the voice quality enhanced significantly and SIPHello can interwork with most of SIP UA.
引文
[1] M Brandl, D. Daskopoulos. IP Telephony Cookbook, 2004
[2] R. Braden, L. Zhang, S. Berson, S. Herzog, S. Jamin .Resource ReSerVation Protocol. RFC2205, September 1997
[3] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler .SIP: Session Initiation Protocol. RFC3261. June 2002
[4] D Crocker, P Overell .Augmented BNF for Syntax Specifications:ABNF. RFC2234, 1997
[5] M Handley, V Jacobson. SDP: Session Description Protocol.RFC4566, 2006
[6] H Schulzrinne S Casner. RTP: A Transport Protocol for Real-Time Applications. RFC3550, 2003
[7] ITU-T Recommendation G.711. Pulse code modulation (PCM) of Voice Frequencies, 1992
[8] ITU-T Recommendation G.729. Coding of speech at 8 kbps using Conjugate-Structure Algebraic-Code-Excited Linear Prediction (CS-ACELP).1996
[9] S. Andersen, A. Duric, H. Astrom. Internet Low Bit Rate Codec (iLBC). RFC3951, 2004
[10] J. Valin. The Speex Codec Manual[S], 2007
[11] ITU-T. H.263 Standard . Video coding for low bitrate communication. 2005
[12] ITU-T. H.264 Standard . Advanced video coding for generic audiovisual services[S]. 2005
[13] ISO/IEC 14496-10 (MPEG-4) AVC. Advanced Video Coding for Generic Audiovisual Services, 2003
[14] ISO/IEC 11172-2 (MPEG-1). Coding of Moving Pictures and Associated Audio for Digital Storage Media at up to About 1.5mbit/S Part 2: Video, 1993
[15] ISO/IEC 138182 (MPEG-2 Video). Generic Coding of Moving Pictures and Associated Audio Information - Part 2: Video, 1994
[16] Y. Kikuchi, T. Nomura, S. Fukunaga, Y. Matsui, H. Kimata. RTP Payload Format for MPEG-4 Audio/Visual Streams. RFC3016, November 2000
[17] C. Zhu. RTP Payload Format for H.263 Video Streams. RFC2190, 2002
[18] 陆其明 . DirectShow 开发指南. 北京:清华大学出版社, 2003
[19] SIPSYS-CS 技术白皮书. 2008
[20] 李洋 . 计算机网络实验教程. 北京:重庆大学出版社. 2008
[21] Counterpath. Eyebeam v1.5 User’s Guide for Windows. 2008
[22] LC-Text: Recommendation Mean Opinion Score (MOS) Terminology[S]. 2003
[23] ITU. Subjective performance assessment of telephoneband and wideband digital codecs[S]. 2001
[24] ITU-T Recommendation P.861. Objective Measurement of Telephone Band (300-3400Hz) Speech Codecs (PSQM). 1996
[25] ITU-T Recommendation G.107. The Emodel, a Computational Model for Use in Transmission Planning. 1998
[26] ITU-T Recommendation G.107. Application of the Emodel:a planning guide. 1998
[27] Damien Stolarz. Mastering Internet Video. USA: Assiaon-Wesley, 2005
[28] (美)卡姆阿洛(Camarlo, G.) . SIP 揭密. 北京:人民邮电出版社, 2003
[29] (芬)卡马里罗 马丁. 3G IP 多媒体子系统 IMS:融合移动网与因特网. 北京:人民邮电出版社, 200
[30] 谢希仁 . 计算机网络(第 5 版). 北京:电子工业出版社, 2008
[31] 求是科技 . Visual C+ 音视频编解码技术及实践. 北京:人民邮电出版社, 2006
[2] R. Braden, L. Zhang, S. Berson, S. Herzog, S. Jamin .Resource ReSerVation Protocol. RFC2205, September 1997
[3] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler .SIP: Session Initiation Protocol. RFC3261. June 2002
[4] D Crocker, P Overell .Augmented BNF for Syntax Specifications:ABNF. RFC2234, 1997
[5] M Handley, V Jacobson. SDP: Session Description Protocol.RFC4566, 2006
[6] H Schulzrinne S Casner. RTP: A Transport Protocol for Real-Time Applications. RFC3550, 2003
[7] ITU-T Recommendation G.711. Pulse code modulation (PCM) of Voice Frequencies, 1992
[8] ITU-T Recommendation G.729. Coding of speech at 8 kbps using Conjugate-Structure Algebraic-Code-Excited Linear Prediction (CS-ACELP).1996
[9] S. Andersen, A. Duric, H. Astrom. Internet Low Bit Rate Codec (iLBC). RFC3951, 2004
[10] J. Valin. The Speex Codec Manual[S], 2007
[11] ITU-T. H.263 Standard . Video coding for low bitrate communication. 2005
[12] ITU-T. H.264 Standard . Advanced video coding for generic audiovisual services[S]. 2005
[13] ISO/IEC 14496-10 (MPEG-4) AVC. Advanced Video Coding for Generic Audiovisual Services, 2003
[14] ISO/IEC 11172-2 (MPEG-1). Coding of Moving Pictures and Associated Audio for Digital Storage Media at up to About 1.5mbit/S Part 2: Video, 1993
[15] ISO/IEC 138182 (MPEG-2 Video). Generic Coding of Moving Pictures and Associated Audio Information - Part 2: Video, 1994
[16] Y. Kikuchi, T. Nomura, S. Fukunaga, Y. Matsui, H. Kimata. RTP Payload Format for MPEG-4 Audio/Visual Streams. RFC3016, November 2000
[17] C. Zhu. RTP Payload Format for H.263 Video Streams. RFC2190, 2002
[18] 陆其明 . DirectShow 开发指南. 北京:清华大学出版社, 2003
[19] SIPSYS-CS 技术白皮书. 2008
[20] 李洋 . 计算机网络实验教程. 北京:重庆大学出版社. 2008
[21] Counterpath. Eyebeam v1.5 User’s Guide for Windows. 2008
[22] LC-Text: Recommendation Mean Opinion Score (MOS) Terminology[S]. 2003
[23] ITU. Subjective performance assessment of telephoneband and wideband digital codecs[S]. 2001
[24] ITU-T Recommendation P.861. Objective Measurement of Telephone Band (300-3400Hz) Speech Codecs (PSQM). 1996
[25] ITU-T Recommendation G.107. The Emodel, a Computational Model for Use in Transmission Planning. 1998
[26] ITU-T Recommendation G.107. Application of the Emodel:a planning guide. 1998
[27] Damien Stolarz. Mastering Internet Video. USA: Assiaon-Wesley, 2005
[28] (美)卡姆阿洛(Camarlo, G.) . SIP 揭密. 北京:人民邮电出版社, 2003
[29] (芬)卡马里罗 马丁. 3G IP 多媒体子系统 IMS:融合移动网与因特网. 北京:人民邮电出版社, 200
[30] 谢希仁 . 计算机网络(第 5 版). 北京:电子工业出版社, 2008
[31] 求是科技 . Visual C+ 音视频编解码技术及实践. 北京:人民邮电出版社, 2006