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Re: [Sip] Call fork



Consult the RFCs 3959 and 3960.

What about muting the RTPs except the last one and as soon as a 200 OK is received you re-open the valid connection. Of course you will expect a media clipping. I think this is the simplest one.

Pavlos


Dean Willis wrote:

On Sep 22, 2009, at 7:48 AM, <sairam.pokkunuri at wipro.com> <sairam.pokkunuri at wipro.com> wrote:

 Hi

 I have a scenario where  Invite is forked by a proxy and it gets multiple 183 containing sdp and all are sending rtp packets as anouncements.What should the b2bua do?
 As we dint know which ua is going to send 200 ok


Congratulations. You've discovered promiscuous early media. That's why I think this part of the protocol spec is broken; early media (driven by PSTN-interworking) combined with forking produces an intractable problem.

Various hacks have been used to address the problem. As far as I know, none of them really work "right", but you might find one that works most of the time for your use case.

I suggest trying the sip-implementors mailing list for advice on this issue:

<sip-implementors at lists.cs.columbia.edu>

Although personally, what I'd probably do is hunt down the developers of the UAs that are all sending 183 with SDP and related RTP and verbally abuse them. Of course, if the UAs are PSTN gateways, there really isn't much else they can do.

--
Dean
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--
Paraskevopoulos Pavlos
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