热门标签： server webrtc voip asterisk android sip sip-server pjsip ios rtp

• web applications - SIP trunk returns ＂SIP2/0 513 Message to big＂. sipjs and oversip

  webrtc

• 纯软的SIP语音服务器API

  以前的语音服务器如呼叫中心、IVR等应用，一般是用语音板卡完成，如三汇和东进的数字卡，模拟卡。如今软交换是趋势，有的运营商甚至可以提供基于IMS的sip中继，再使用语音卡就不合适了。纯软的语音系统，部署起来也简单，只要弄一台普通的PC或服务器就可以，录音也存放在该机器上。 我们自

  语音

• sip 基本概念

  基本概念 Session:session简单的说就是一次通话，从摘记拨号开始到最终的挂机之间所有的sip消息都属于一个session,他们具有相同的call-id。 Dialog:?基于peer-to-peer，描述了两端的user agents在一段时间内的关联。Dialog用

  sip

• JAIN SIP on android with java.lang.reflect.InvocationTargetException

  jain-sip

• linux - tshark SIP protocol display collapse

  tshark

• rfc - Why is 200 OK retransmission handled by UAS in SIP Protocol?

  sip

• sip - Freeswitch with Skype Connect external profile keeps timing out and disconnecting

  skype

• p2p - What is the need for the SIP RE-INVITE with regards to ICE?

  nat

• restcomm call between webRTC client and a normal SIP client ends with 480 message

  webrtc

• SIP VoIP calling in blackberry

  voip

• How to implement audio ring for SIP outbound calls

  call

• sip - how to merge two gsm and voip calls on android phone

  gsm

• SIP to WebRTC call. SIP sdp body/message content anomalies

  call

• resteasy - JBoss REST with SIP Servlet - Class Not Found due java.lang.LinkageError

  rest

• How to notify SIP client when there is an incoming call on another phone

  sip

• skype - Set a SIP sjphone as default

  skype

• Open source solution for SIP based video streaming server?

  server

• ios - Cannot reach SIP server

  nat

• sip - application unable to access internet on device while working properly on android emulator?

  user-permissions

• sip - Does liblinphone in Android support SUBSCRIBE with dialog event?

  linphone