Re: [reSIProcate-users] Establishing a direct RTP session between 2SIP clients

[Byron Campen <bcampen@xxxxxxxxxxxx>]

	Ah, that would do it then! Of course, this will only work if both  
endpoints handle reINVITEs correctly, but that isn't asking for much.

Best regards,
Byron Campen

> 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
> >
> >
> >
> > _______________________________________________
> > resiprocate-users mailing list
> > resiprocate-users@xxxxxxxxxxxxxxx
> > List Archive: http://resiprocate.org/archive/resiprocate-users/

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