[Asterisk-Dev] SIP bug and Polycom phones

[Clif Jones]

See below.

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 8.1.1.7, changing the branch number for each 
>new transaction is the correct behaviour. I guess the question is, are 
>reinvites new transactions? Polycom thinks so.
>  
>
Re-INVITES are new transactions on the same SIP dialog.
A new transaction gets a new branch ID which is used to support forking 
and several other things.

>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 {
>		// BEGIN POLYCOM SPECIFIC CODE
>        	p->branch++;
>        	snprintf(p->via, sizeof(p->via), "SIP/2.0/UDP 
>%s:%d;branch=z9hG4bK%08x", inet_ntoa(p->ourip), ourport, p->branch);
>		// END POLYCOM SPECIFIC CODE
>		reqprep(&req, p, "INVITE", 0);
>    	}
>
>... the rest of the function follows.
>
>Thanks,
>
>Christian
>
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>  
>