Hi,
I am having a strange problem binding audio ports during media transfer.
I am parsing the audio port number which I get as SDP from my SIP Server. I
use this port number for sending my RTP Packets. On the other hand when I
use the audio port number which I have sent through my INVITE to the Server,
I cannot even establish a connection. But when I use the standard SIP port
5060 as the port to bind my local address, I have got no problem in sending
the RTP Packets to the SIP Server. I am really surprised about this
behavior. Can SIP professionals shed some light on this.
For example my *SIP* *INVITE *message looks in this manner:
"INVITE sip:[email protected] SIP/2.0\r\n"
"Via:SIP/2.0/UDP
192.168.x.000:5060;branch=z9hG4bKdg18\r\n"
"Max-Forwards: 70\r\n"
"To: server <sip:[email protected]>\r\n"
"From: User<sip:[email protected]>; tag
= 76341\r\n"
"Call-ID:
[email protected]@192.168.x.x\r\n"
"CSeq: 1 INVITE\r\n"
"Contact: <sip:[email protected]>\r\n"
"Content-Type: application/sdp\r\n"
"Content-Length:142\r\n";
"\r\n"
"v=0\r\n"
"o=User 53655765 2353687637 IN IP4
192.168.x.000\r\n"
"s=-\r\n"
"c=IN IP4 192.168.x.000\r\n"
"t=0 0\r\n"
"*m=audio 12856* RTP/AVP 0 8\r\n"
"m=video 51372 RTP/AVP 98 49\r\n"
"a=rtpmap:0 PCMA/8000\r\n"
"a=rtpmap:98 H263-1999/90000\r\n";
here I am sending information to the Server saying that for the media part I
communicate on port number *12856*.
on the other hand if I do this
*#define AUDIOPORT 12856*
When I use like this *ClientAddr.sin_port = **AUDIOPORT**; *
I cannot send any data to the Server (which I can see on my Wireshark
Capture)
*#define SIPPORT 5060*
When I use like this *ClientAddr.sin_port = SIPPORT; *
my client gets bind to the server and the transactions takes place and I can
see the connection establishment and transfer of RTP packets
Am I doing some mistake in the initializations.
Regards
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