>>My question is, how can I ask caller to keep sending message to my
UNIX box?
I think this is OK.and is possible.
In the SDP information while sending with the Invite include the Port
number and IP address of the Unix Box
Than the other entity after the INVITE-OK-ACK will establish media path
with parameters in the SDP. (which are actually of your UNIX Box)
In SDP for media description we have (mainly these),
m= (media name and transport address)
c= (connection information ) // This is actually optional;
So while tearing down the connection (when Bye is send) you should take
care to stop the rtp packets and closing the RTP session.
Any comments??
Thanking you,
Regards,
Sreeram
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jenning
zhang
Sent: Thursday, January 22, 2004 1:01 AM
To: [EMAIL PROTECTED]
Subject: [Sip-implementors] If SIP and RTP part using different IP
address
Hi there,
I'm new here and new in SIP application development. I'm developing a
SIP voice mail application. The SIP is running
on a Solaris box (IP1) and the media part (RTP and voice stuff) is using
a third party PCI card (which has a separate
IP address IP2). So here comes a problem. SIP on the unix box receives
INVITE and sends 100 and 200 back, then
the caller (a SIP phone) sends the rest of the messages (ACK, BYE)
directly to the IP address pointed by SDP instead
of Contact. So in my case, the rest of the messages are sent to the PCI
card, but the CARD has no SIP on it.
My question is, how can I ask caller to keep sending message to my unix
box?
Thank you very much
Jenning
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