SIP协议实现的软电话功能例子(仅供参考)

客户端使用NetMeeting，服务端使用mfc.exe可以获得比较好的效果-clients use NetMeeting, The server can use mfc.exe fairly good results

VOIP(H323)源码，实现了G723.1 G729 G728 G711等音频算法，可以和NetMeeting互通-VoIP (H323) source, Implementation of G723.1 G729 G728 G711 audio algorithms, etc., NetMeeting and can exchange

rtp模块， 包括rtp协议报文组包，解析，功能独立，需要自行添加rtp的session管理-rtp module, including the rtp protocol packet group package, analytical, functional independence, the need to add the session management rtp

GSM 06.10 13kbit/s RPE/LTP 语音压缩算法 -GSM 06.10 13kbit/s RPE/LTP voice compression algorithm

nettalk,个人觉得非常不错的网络电话，集成视频，声音和文本聊天。提供源码，大家共同学习

在ip电话或者视频通讯开发中，常常需要穿越nat，stun是个不错的选择-In the ip phone, or video communication development, often through the nat, stun is a good choice

eXosip-2.3.0.1：eXosip最新版本，配合Osip使用-eXosip- 2.3.0.1 : eXosip latest version, with the use of Osip

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

SDP2327，中文版，学习sipHello的同学可以参考一下-SDP2327