[Live-devel] playSIP crashed after getting 200 OK for invite

[Ross Finlayson]

> Because there are some limitations within my RTP server and I would like to replace it with RTP module from Live555 package  

What you’re describing is probably possible, but would be quite complex, especially for someone who’s inexperienced in C++.  (I assume that you’ve already implemented the appropriate SIP commands in your server.)

> 
> I am newbie in LIVE555 and also on learning edge of C++. 

> Could anyone shed any light on this how can I make use of RTP module from Live555 package in my current project?
> How can I by pass RTSP server and directly call RTP library?

Your server would need to create appropriate “RTPSink” (for RTP) and “RTCPInstance” (for RTCP) objects, in the same way that our RTSP server implementation already does (using the “OnDemandServerMediaSubsession” class).  Unfortunately, this will require quite a bit of understanding of the LIVE555 code, and C++ expertise; it’s not really a task for someone who’s just learning the language.  You also can’t expect a great deal of support ‘for free’ on this mailing list.

Therefore, you might find it better to look instead at other existing Open Source SIP servers/PBXs - e.g., Asterisk.


Ross Finlayson
Live Networks, Inc.
http://www.live555.com/