sip

I am creating a voip call based project with Linphone and I have also successfully build and run the Linphone project and successfully run audio and video call. Now I am integrating Linphone in my own project and I am facing many problems and issues with this. I have used some following links for help but nowhere are complete instructions. Can anyone provide me the complete running steps for this- http://shallwelearn.com/blog/build-linphone-for-iphone-and-ipad/ Integrate Linphone app to my iOS app How to integrate Linphone into an existing project (SIP in IOS) http://www.linphone.org/technical

I am able to implement Facebook and Gmail chat with the help of XMPP Framework in my iPhone app. Wanted to know if its possible to implement VOIP(SIP) in a similar manner using XMPP. UVM You can use jingle framework.This is what jingle wiki says: "Jingle is an extension to the Extensible Messaging and Presence Protocol (XMPP) which adds peer-to-peer (P2P) session control (signaling) for multimedia interactions such as in Voice over IP (VoIP) or videoconferencing communications. It was designed by Google and the XMPP Standards Foundation. The multimedia streams are delivered using the Real-time

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: I have a windows desktop application, made by my mobile network provider, that does all kind of things with SIP: call, send message, etc. Screenshot of how does this app successfully send MESSAGE (the last 4 lines): MESSAGE request, from desktop application, is sent as (4th line from behind) : MESSAGE sip:FROM@DOMAIN SIP/2.0 Via: SIP/2.0/UDP LOCALIP:2112;branch=z9hG4bK-d8754z-905183245f478c76-1---d8754z-;rport Max-Forwards: 70 To: "TO"<sip:TO@DOMAIN> From: "FROM"<sip:USERNAME@DOMAIN>;tag=63088d09 Call-ID:

I've managed to set up a SIP call using the JAIN-SIP API for Java. Now I would like to stream some video once a dialog has been established. I've read that this is possible with SDP and RTP, and I've found multiple examples on how to define a SDP/RTP body in a SIP packet. But once you have negotiated capability etc. on nodes, how do you actually start the RTP stream? Do you start an RTP streaming server outside or inside your Java application? If so, how? What is the link? In what I'm able to find online, nodes just "start exchanging RTP packets". Thank you. You need an RTP stack. As you are

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: How to build and compile PJSIP using xCode and run the sample code IPJSUA? 回答1: First, you need to open the terminal. Terminal can be found at Applications->Utilities->terminal. After you open the terminal, make sure you point to the desktop to make it easier to get the data folder. just type : cd Desktop Congrats, you already at your desktop. Now continue to type this. svn co http://svn.pjsip.org/repos/pjproject/trunk pjproject That code means you download the pjproject from the website to your desktop. After you finish download the PJSIP,

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: I am building a basic SIP UA. I am sending the following INVITE, as seen in Asterisk console (only headers relevant to authentication are shown): INVITE sip:104@192.168.1.92 SIP/2.0 From: "110"110>104>110> 104>110> 110>104>110>110>104>110> 110>

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 选择语言 中文(简体) 日语 英语 中文(繁体) 由 翻译 强力驱动 问题: Here I'm developing an application using native android in ubuntu 14.04LTS eclipse adt for android. Build, configure, and make everything went perfect. After that I imported pjsua2 sample app into eclipse, I then added native library support but when I run on emulator it throws an error. The error log, 10-25 06:12:09.489: E/AndroidRuntime(1571): FATAL EXCEPTION: main 10-25 06:12:09.489: E/AndroidRuntime(1571): Process: org.pjsip.pjsua2.app, PID: 1571 10-25 06:12:09.489: E/AndroidRuntime(1571): java.lang

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: I have created a sip trunk from One Asterisk(version 11.2.1) say 'A' server to another Asterisk server(11.7.0) say 'B', and I am getting sip response 200 ok. But when I start calling on a DID on Asterisk A then the call is being routed to Asterisk 'B' and After 38 seconds call has been disconnected showing following warnings : Retransmission timeout reached on transmission 11bc71e029119e5877806ed40fcde691@111.xxx.xxx.xxx:5060 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: I see above log message when initiating SIP call from Android app using Android SIP SDK. Here's the line which causes it: mSipManager.makeAudioCall(mSipProfileLocal, mSipProfilePeer, listener, 20); I don't think there's anything wrong (at least related to the above message) in that line. But anyway, after that method is called, I see recursive attempt to load library "/system/lib/librtp_jni.so" And SIP call never gets established. Also, I don't receive any error messages/exceptions - nothing. Have any idea? Thanks in advance. Edited: Btw,

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: I'm trying to build a simple SIP application using JAIN SIP 1.2 and the NIST implementation. I'm using JavaSE1.7 with Eclipse as my IDE. The problem: I am getting javax.sip.PeerUnavailableException when trying to construct a SipStack object. My main class: package net.bezeqint.sip.enp; public class ListenerMain { public static void main(String[] args) { try { System.out.println("Creating ExampleListener..."); ExampleListener listener = new ExampleListener(); } catch (Exception e) { e.printStackTrace(); System.exit(-1); } } } My problematic