Re: [reSIProcate-users] Establishing a direct RTP session between 2SIP clients
As far as I know, it is possible. From memory, it caused us problems in
our environment because we didn't want it to re-invite .... there's more
detail here...
http://www.voip-info.org/wiki-Asterisk+SIP+canreinvite
Steve Coule
-----Original Message-----
From: resiprocate-users-bounces@xxxxxxxxxxxxxxx
[mailto:resiprocate-users-bounces@xxxxxxxxxxxxxxx] On Behalf Of Byron
Campen
Sent: 09 October 2007 15:49
To: Megha Saini
Cc: resiprocate-users@xxxxxxxxxxxxxxxxxxxx
Subject: Re: [reSIProcate-users] Establishing a direct RTP session
between 2SIP clients
Asterisk doesn't support this sort of thing, as far as I know.
This
is because it is functioning as a B2BUA, and is terminating the media
on both sides. I don't know if Asterisk can be configured to function
as a proxy (like SER or repro).
Best regards,
Byron Campen
> Hi..
>
> Using reSIProcate, I am able to establish a SIP session between two
> user agents A and B using the Asterix server. The two use agents
> register on the server and an INVITE message is sent containing the
> correct credentials followed by 200 OK and ACK responses.
>
> However, now I would like to establish a two-way direct RTP session
> between the clients A and B, i.e. without passing the media through
> my server Asterix.
>
> Can anybody please tell me if it is possible to create a direct RTP
> session from A to B after establishment of SIP session between
> them. In such a case, I also need to get the current IP address of
> the callee (not the mnemonic identifier) so that I can send the
> media directly to my client without sending it to the server?
>
> Looking at the logs, I observe that in the SIP session, I get the
> address of the server in the Via field of the SIP header and not
> the IP address of the callee. Also, in SDP, I dont get the callee's
> IP address. Can anybody tell me that how should I get the IP
> address of the callee if I have to establish a direct RTP session
> between the two clients.
>
> Does it also depends on the server that I am using? I am using
> Asterix. Is it possible to create a direct RTP session with that or
> do I have to send the media only through the server everytime? I
> guess that it might create an extra overhead if I pass the media
> through the server for every call.
>
> I have read that when a SIP server like SER receives a message, it
> can decide that it wants to stay in the loop or not. If not, SER
> will provide the user agents with the information they need to
> contact each other and then SIP messages will go directly between
> the two user agents.Is it possible with Asterix too?
>
> I have a deadline of the project. Any help would be highly
> appreciated.
>
> Regards
> Megha
>
>
>
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