process of connecting RTP with SIP via SDP & land lines

Posted by TacB0sS on Stack Overflow See other posts from Stack Overflow or by TacB0sS
Published on 2010-05-23T09:05:06Z Indexed on 2010/05/23 9:10 UTC
Read the original article Hit count: 363

Filed under:
|
|

Hello to everyone,

I have a problem with starting a media session and to combine it with my SIP client. I've designed a recursive SIP client that reuse the same request template to send the next requests to server, according to the acceptable sequences noted in the RFC's, and examples that I read. as far as I could tell the SIP part is working fine registers to server invites, and authenticates. I didn't complete any calls to clients yet because of the content header needs to be filled up (which I didn't yet so I get a 503 from the server which is OK I guess).

for a long time I didn't know where to start with the media session, and slowly learned how to use the JMF and I've constructed an object that handles RTP transmitting, now I'm standing at the cross road, on the one hand I have my SIP signaling but it needs the SDP content header to complete the invite, and on the other I have the RTP which is knows how to p2p.

For me to complete my design I require your help with the following questions:

  1. Is there an easy//a simple//an implemented way to convert the Audio/Video format from the JMF into SDP media headers? or even a generator that I would input all the parameters for the content header, and it would generate a content header fast, or do I have to implement this myself?

  2. Once I've finished constructing the SDK and the SIP is up and running and I get an OK response from the server (after ringing and all), how do I start the media session? do I connect p2p according to caller details I send in the SIP invite?

  3. If 2 is correct, then how does a connection to land lines would be? does land lines knows that once they send an OK back to server they listen/start RTP session on a specific port?

Or did I get everything wrong? :-/

I really appreciate any help I could I get, I looked every where for answers but they are not clear, they ignore question 2 as if it was an obvious thing, but for me it just isn't.

Thank in advance, Adam Zehavi.

© Stack Overflow or respective owner

Related posts about sip

Related posts about rtp