/csip

CSipSimple 是个款通用的支持SIP协议的互联网电话软件,可以在支持andriod的平板,手机上使用。支持语音编码: G.711 aLaw/uLaw, G.722.1, G.722, SPEEX, SPEEX-WB, AMR-WB, GSM, iLBC, G.729. 支持STUN, Outbound proxy server, Qos,VAD,AEC,AGC,CNG等众多功能和标准。

Primary LanguageCGNU General Public License v3.0GPL-3.0

CSipSimple

CSipSimple 是个款通用的支持SIP协议的互联网电话软件,可以在支持andriod的平板,手机上使用。

带编译环境虚拟机

配置好编译环境的virtualbox虚拟机:
链接:https://pan.baidu.com/s/1PKGCuWT_XHPO7JR5ya_riQ

虚拟机中代码路径:
/home/build/code.Proj

备份官方svn仓储代码路径:
/home/build/backup

虚拟磁盘格式为vmdk,也可以挂载到vmware或者parallels虚拟机中使用。


本源码仓储由官方SVN转成git,并由ZenCodex 维护,如果有问题,请邮件联系 v@yinqisen.cn

**CSipSimple爱好者QQ交流群:13945450


This project will allow native sip for android device.

It relies on the pjsip sip stack project.

Supported features :

SIP for calls and instant messages
Android integration with rewriting and filtering rules
Codecs : pcmu/a (aka g711u/a); speex; g722; gsm; iSAC; SILK; G729; AMR (depending on device) and as extra plugin : OPUS; g726; g722.1; codec2
Echo cancellation (with various backends : webRTC, speex, simple)
Auto speaker/earpiece option
API provided for other apps plugins.
Conference, transfer.
Secure call with TLS transport for SIP and SRTP or ZRTP for media.