sip

I have the FFMPEG streaming baseline h264 video, which I have to encapsulate in RTP and send to SIP phones for their decoding. I am using Linphone with the h264 plugin for Windows and Mirial for the decoding progress. However, sometimes I get a huge frame size (3Kb ~ 9Kb) from the FFMPEG, which obviously doesn't fit in the MTU. If I send these frames "as is" and trusting IP fragmentation feature, some phones are able to play it well enough, but others choke and can't decode the stream. I think this is because the stream is not compliant with the RFC 3984 that specifies that packets that don't

在大部分的企业客户的电话呼叫业务中，特别是从运营商到企业IPPBX端的呼入业务中，有很多不同的呼叫涉及了多种SIP流程的操作，而且其流程和实际的IPPBX，代理和SIP终端存在着非常密切的关系。排查这些技术问题耗费相当多的时间。另外，因为越来越多的用户开始使用基于开源的软交换平台和媒体服务器（例如，Asterisk或FreeSWITCH，Kamailio等），用户更容易获得技术产品，因此，他们更容易接触到企业通信平台技术，导致其入门门槛也相对比较低，技术人员可能完全不了解系统底层的流程。而且不幸的是，在实际使用过程中，很多技术人员也仅仅停留在通过系统界面配置一个呼叫业务流程，他们根本没有了解和关注真正底层的呼叫流程和其细节，而且他们对真正的SIP消息之间的互相通信过程可能并不是非常熟悉。笔者发现，其中一个原因是他们没有太多学习渠道获得一些非常直观和权威的可参考的示例。 　　大家经常谈论SIP呼叫业务，但是，究竟哪些SIP呼叫业务是企业用户所要求的？ 关于SIP业务呼叫，RFC5359对18个最常用的SIP业务呼叫流程给出了完整的SIP流程图例，这些呼叫业务为企业用户解决方案部署提供了一个比较权威的参考。因此，笔者希望通过此文章完整列出所有18个关于SIP呼叫业务的SIP流程和其相应的图例说明，并且加以适当讨论和说明来解释这些呼叫功能中可能出现的问题或部署时应该注意到地方

I have a Twilio account which has a number (let's say 8881231234), and I have Asterisk box. I'd like to use Twilio as an Asterisk trunk to be able to make calls at their rates and receive calls from that number on my Asterisk. I haven't found any specs to interconnect Asterisk with Twilio. Is it possible to set up Asterisk so that every outgoing call is routed through Twilio and have the calls on my 8881231234 number ring on my SIP phone? user2152758 Twilio Seems to offer SIP Trunking now. g3rv4 As it was well pointed out below, Twilio now DOES WORK as a SIP trunk, you can visit https://www

I have this challenge to build an sip softphone using c# or .net technologies. Please guide me the technology, requirements and specifications that is needed to build such. Possible requirements: Supported codecs: G.722.1, G.723.1, G.726, G.728, G.711, G.729, G.723.1, iLBC,: G.711 (A and m-law),G.729A/B/D/E, AMR, GSM 6.10/EFR, iLBC, Speex Voice: Real time Quality Monitoring (MOS) 3 independent phone lines Auto-Answer/Do Not Disturb Call Forwarding Full Duplex Audio Recording compress the audio data. Fully SIP compatible softphone/dialer for PC2Phone and Mobile2Phone (iPhone, Nokia N95)

My objective: I want to use softphone(3CX phone) register with asterisk server, and make call to the server and asterisk act as a server to automatically response something, like play a song. How i did: I installed asteriskNow using virtualbox, and registered the softphone by setting exntension for my SIP device (extension 333). And i write a dialplan in etc/asterisk/extensions.conf . The dialplan is : [incoming] exten =>s,1,Answer() exten =>s,n,Playback(dir-intro-oper) exten =>s,n,Hangup() I want any incoming call to server, the server will automatically answer, and play a pre-defined voice

i am developing a SIP application for making and receiving a call and i want to add the G729 codec in my application. currently i am doing analysis on open source project SipDroid . if i want to make that application to support G729 codec how to do that? there is a different codecs configuration file in org.sipdroid.codecs package.how do create the this kind of .java file for G729 codec? Any suggestion and response will be appreciated. Log message of asterisk Found RTP audio format 101 Found audio description format telephone-event for ID 101 Found RTP video format 103 Found video description