[Asterisk-Dev] SIP bug and Polycom phones

Christian Hecimovic checimovic at qworks.ca
Thu Dec 11 17:43:00 MST 2003

Thanks for the info guys, I've submitted the patch. 


On Thursday 11 December 2003 11:02, Christian Hecimovic wrote:
> Hi,
> Polycom phones track the current SIP transaction with the branch number in
> the Via header. When Asterisk does a reinvite to set up a native bridge, it
> does not change the branch number - it's still the magic cookie plus some
> network info plus an unchanging random number stored in sip_pvt.branch. In
> short, the branch number stays the same from one invite to the next.
> To get reinvites to work, I made a very simple fix in
> transmit_reinvite_with_sdp(). I incremented the branch number and recreated
> the via field of the sip_pvt struct before making the call to reqprep().
> Now Polycom phones perform native bridging; sniffing the network shows that
> the RTP streams do not touch the server.
> According to RFC 3261, Section, changing the branch number for each
> new transaction is the correct behaviour. I guess the question is, are
> reinvites new transactions? Polycom thinks so.
> Before I submit this little patch, I would really like to hear from anyone
> who thinks this may destabilise Asterisk with other phones, who assume the
> branch number will stay the same between invites. I do not have any other
> phones to test with. The code is as follows:
> static int transmit_reinvite_with_sdp(struct sip_pvt *p, struct ast_rtp
> *rtp, struct ast_rtp *vrtp)
> {
> 	struct sip_request req;
> 	if (p->canreinvite == REINVITE_UPDATE)
> 		reqprep(&req, p, "UPDATE", 0);
> 	else {
>         	p->branch++;
>         	snprintf(p->via, sizeof(p->via), "SIP/2.0/UDP
> %s:%d;branch=z9hG4bK%08x", inet_ntoa(p->ourip), ourport, p->branch);
> 		reqprep(&req, p, "INVITE", 0);
>     	}
> ... the rest of the function follows.
> Thanks,
> Christian
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