Hi, I have a SIP application (written in C) and my RTP server also in C. Below is how they work
From SIP client side 1. Create a socket connection to RTP server using RTP client library 2. Then send CREATE_CMD to RTP server to create rtp media session with SDP parameters for this call 3. Send INVITE 4. After call is established send PLAY_CMD/RECORD_CMD to RTP server to play/record a wav file Because there are some limitations within my RTP server and I would like to replace it with RTP module from Live555 package 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? Any help will be greatly appreciated. Thank you, Sak -----Original Message----- From: live-devel [mailto:[email protected]] On Behalf Of Ross Finlayson Sent: 08 February 2016 15:02 To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] playSIP crashed after getting 200 OK for invite Thanks for the note. I’ve just released a new version (2016.02.08) of the code that should fix this. Ross Finlayson Live Networks, Inc. http://www.live555.com/ _______________________________________________ live-devel mailing list [email protected] http://lists.live555.com/mailman/listinfo/live-devel _______________________________________________ live-devel mailing list [email protected] http://lists.live555.com/mailman/listinfo/live-devel
