If we have to send RTP media from one client to another when a session has been established between them using SIP and SDP negotiation has been done either in either INVITE or ACK, what parameters needs to edited in header when relaying media through a server sitting in middle of the clients.
1
votes
From your question it's really hard to answer what you are specifically looking for. Usually a Via header is added if you want your call to be passed through certain gateway.
– rakib_
Only for media relaying. For SIP messages, we do change via, record route etc but when we have to send RTP and RTCP packets through a b2bua, what parameters do we must rewrite(ssrc etc?
– Hooli
Well, that is usually done during by SDP modification. You can read code from projects like OpenSIP's rtpproxy modules and rtpproxy itself. You will know how it's done.
– rakib_
Alright, I will look at that!
– Hooli
2 Answers
1
votes
The only time the SDP needs to be modified in a B2BUA is if you have a need to do something with the media flowing between the endpoints involved with the call. Typically the endpoints handle the SDP processing but if you need to do something like transcoding of the media using a media resource controlled by the B2BUA, the media server typically takes care of the SDP modifications.
For example, if you are using a SIP servlet container that supports JSR 309 for media server control to build your B2BUA, SDPs are exchanged between the SIP container and the media server over MSML. The B2BUA usually does not directly modify the SDP. JSR 309 is the API for communicating with a media server. Media servers like Dialogic XMS or the Radisys MRF take care of all the SDP manipulation and all the B2BUA needs to worry about is passing around the SDPs being processed by the media server.
0
votes
You need well defined SIP proxy, and handle it like described here: https://www.voip-info.org/wiki/view/RTPProxy
