IP电话编程

在开发sip通话或者h323的时候，都想需要作认证，所以这里给出本人开发的一个很小的digest认证流程的小例子。-call in the development or sip h323 time, and they all want to require certification So I developed here is a very small digest authentication process small examples.

用C＋＋语言写的开源SIP协议栈源码 编译环境为vs.net 2003-C++ language to write open-source SIP protocol stack source compiler environment vs.net 2003

无线环境下的视频传送信道模拟软件-wireless environment video transmission channel simulation software

VOIP program in VC++, video, voice and other features

在开发 VoIP 系统时，主要考虑的一个方面就是连接到模拟电话的接口。设计人员必须了解 PSTN 中存在的电话接口要求，因为它们必须在 VoIP 系统中得到支持。本文将重点介绍连接到标准 POTS 电话的两种最常用接口：Foreign eXchange Subscriber (FXS) 与 Foreign eXchange Office (FXO)。重点说明了设计人员在为 VoIP 小区网关的模拟电话接口提供支持时可能面临的某些挑战。-In the development of VoIP system, one aspect of the main considerations is connected to the analog telephone interface. Designers must understand the existing PSTN telephone interface requirements, because they are in the VoIP system must be supported. This article will focus on the standard POTS telephone connected to the two most commonly used interface: Foreign eXchange Subscriber (FXS) and Foreign eXchange Office (FXO). Focusing on the design staff in the Residential Gateway for VoIP analog telephone interface may provide support to some of the challenges faced.

introduce to sip h323

可以用vc编译的一个sip 客户端,使用一个基于osip,exosip,ortp和mediastreamer的高层库.有该库的使用文档-Vc compiler can use a sip client, based on the use of a osip, exosip, ortp and high-level Treasury mediastreamer. The use of the library documents

主要通过对SIP协议的研究，根据这个协议的特点，设计出来一个基于SIP电话会议系统。用户可以通过WEB方式登录到预先建立好的会议室，进行电话会议。-primarily through the SIP, according to the characteristics of this agreement, designed a SIP-based telephone conferencing system. Web users can log on to advance ways to establish good meeting rooms, conference calling.

IP电话源程序，在Unix环境下编译通过，基于h.323协议。-source IP telephony, in the Unix environment compiler, based on the H.323 agreement.

2.0说明 1.0支持系统的录音多种格式,有一定的延时,没有过滤杂音功能. 2.0已取消支持系统的录音多种格式,使用PCM采集数据G711A压缩格式(8000HZ单声道16位格式录音每秒以8K完成数据,16000HZ单声道16位格式录音每秒以11K完成数据,音质相当好),延时降低到最小100-500MS以内,不会随时间增加而增加延时(如果是说话测试一直保持200MS的延时,如果是用播放歌曲来测试,有自动校正延时功能,恢复成200MS的延时,恢复过程中不会中断歌曲的播放,只是小小加快唱歌的速度,听觉不会觉察出来),加入了过滤杂音功能. 2.0安装 将ACMG711.dll复制到系统文件夹里 再安装控件 -2.0 1.0 Note recording system to support multiple formats, a certain delay, no noise filtering function. 2.0 support system has been abolished recording formats, the use of PCM data acquisition G711A compression format (8000HZ 16 mono recording format to 8K per second complete data 16000HZ mono recording format 16 to 11K per second complete data quality is very good), Delay reduced to the minimum 100-500MS within not with time delay increases (if word has been tested 200MS delay, If it is used to test play the songs, the automatic correction function of delay, 200MS resume into the delay, the recovery process will not be interrupted play songs, singing just a small speed up