sip

问题 I need to test an application that processes SIP requests. For now, I want to test the performance of the application, so I need a way to generate a big number of SIP requests. I know there are tools for this (like SipP), but I don't know what is the maximum number of requests that a single computer can really send in a particular time interval. I never done this type of test, i need help. Thanks 回答1: Well sipp can generate requests pretty quickly and if you're testing call set up and tear

问题 Hi I need to implement something like SIP phone but with a 'classic' SIP without WebRTC. Most JS libs focus on SIP over websockets and WebRTC, but in my infrastructure, I do not have WebSockets. There are libs like JsSIP even with support for WebSockets in Node.js. Maybe I should solution is to use software like webrtc2sip? But I fill this is little overcomplicated. Some better solution? 来源： https://stackoverflow.com/questions/46559744/simple-sip-phone-in-nodejs-without-webrtc

问题 I'm devlopping a SIP application and whant to launch the normal incoming call interface when having an incoming call from a SIP client. Is that possible ? Thank you very much. 回答1: I am sorry, but it looks like there is nothing as standard incoming call interface in Android (at least not until API level 10). The screen that you see on incoming call is part of the native PhoneApp. More specifically, it is handled by the class InCallScreen.java from the dialer source code. That Activity has no

问题 I'm trying to extract tag from the From: header of a SIP messages. My regex: ^(From:|f:)((?!\\n\\w).)*;[ ]*tag[ ]*=[ ]*([[:alnum:]]*) RFC 3261 allows multi-line headers. This new line should start with a whitespace. But i have a problem with multi-line headers. If the tag is on a new line, my regex is not working. Example correct SIP Message: INVITE sip:13@10.10.1.13 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.99:5060;branch=z9hG4bK343bf628;rport Contact: <sip:15@10.10.1.99> Call-ID:

问题 In extconfig.conf they have mentioned that "However, note that using dynamic realtime extensions is not recommended anymore as a best practice; instead, you should consider writing a static dialplan with proper data abstraction via a tool like func_odbc." 1) Why asterisk is not recommending dynamic realtime extensions? 2) How to do static dialplan with data abstraction using tool liek func_odbc? My requirement is having have more extensions (in this case mobile number) coming up, how can I

问题 I've been coding a SIP client using the Android SDK's native SIP libs. For some reason I cannot get my account to register with the server. Here are the testing grounds: Linux Mint 17 XFCE running a Kamailio Server(MySQL and TLS enabled). Linux Mint 17 Cinnamon running Android Studio (0.8.6). Asus Google Nexus 7 (2nd gen). All of the above are on 192.168.1.xxx The server and account info has been tested with a working SipClient (ZoIPer). The port 5060 has been tested with a telnet command

问题 I have configured Asterisk 13.13.1 with PJProject 2.5.5 and enable PJSIP as SIP driver (without compiling chan_sip). I have the fully configured system and it's working but I have some problems with incoming calls. I have few numbers connected with my host and when I calling from any public number I noticed this info on asterisk remote console: [Feb 24 14:27:16] NOTICE[5291]: res_pjsip/pjsip_distributor.c:525 log_failed_request: Request 'INVITE' from '"zzzzz" <sip:zzzzz@192.168.34.1>' failed

问题 What are the available API or existing technologies on android that can help one make call using the internet, more like VOIP. I checked out documentation online including this . Kinda confused where to start. Can anyone help with a better guideline ? Thanks. 回答1: PJSIP is one of the better media libraries available on Android. you may want to google the pro and cons of this vs the native android SIP Api and based on your project pick one over the other. There are a few others, but PJSIP is

问题 With sipster/pjsip sucessfully installed I would like to follow through the basic sipster example and record the sound from a sip connection to that sip server. Now if I run this on a local arch linux machine the sound gets recorded fully, but if I do the same on a docker machine, that I start with docker run -p 5060:5060/udp -it myContainer the SIP connection works, but there is no sound recorded. A friend told me that SIP uses RTP for sound transport and that this protocol binds UDP ports

问题 I had successfully registered my sip client in pjsua sample app and is receiving incoming calls over UDP. But whenver I try to make an outgoing call it get disconnected with 403 FORBIDDEN error. And the connection created is TCP! I verified the destination uri. it is in the format as sip:mobile-number@sip-server-ip:port here is my makeCall code : public void makeCall(View view) { //some code here.. MyCall call = new MyCall(account, id); CallOpParam prm = new CallOpParam(true); try { call