That makes a lot of sense. Do you think is there anything I could do on my side to solve this? Maybe manually fix the RR?
This client has some kind of link balancing, so public IP may change at any time. This must be the reason he hasn't set it. Thanks in advance! Em 22/04/2016 7:40 PM, "Bogdan-Andrei Iancu" <bog...@opensips.org> escreveu: > Hi Daniel, > > Actually, on the A side, you have another proxy ( see the Record Route > with 172.20.17.11 in the INVITE). OpenSIPS tries to send the BYE to the RR > header, but that is private. A SIP proxy, if sending traffic to public > Internet, should not use at all private IPs. > > Bottom line, the broken link in your scenario is the 172.20.17.11 proxy > before your opensips. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 23.04.2016 00:12, Daniel Zanutti wrote: > > Hi > > I'm facing an strange issue when my Opensips instance hangs up a call, > generating BYE to both sides (timeout on dialog module or rtpproxy). The > BYE is sent to both sides but A side is behind NAT and the BYE is sent to > the local IP address and not to the public one. > > See trace bellow: > > Customer -> Opensips > > *U 200.200.200.200:27923 <http://200.200.200.200:27923> -> > 199.199.199.199:5060 <http://199.199.199.199:5060>* > INVITE sip:551133333...@plat.test.com SIP/2.0 > Record-Route: <sip:172.20.17.11;lr;ftag=7db6f42e;did=769.5ee54854> > Via: SIP/2.0/UDP 172.20.17.11:5060;branch=z9hG4bKb99a.3c0442d4.0 > Via: SIP/2.0/UDP 172.28.0.12:57744 > ;received=172.28.0.12;branch=z9hG4bK-524287-1---08559406a5e9137b;rport=57744 > Max-Forwards: 68 > Contact: <sip:100111@172.28.0.12:57744;rinstance=b4a1be0f56d73cfd> > <sip:100111@172.28.0.12:57744;rinstance=b4a1be0f56d73cfd> > To: <sip:551133333...@plat.test.com> > From: <sip:100111@172.20.17.11>;tag=7db6f42e > Call-ID: 79961NjFjMjY2MDhiYjhhNDYyN2UyMWZiMzFiNWM5NzBkNTM > CSeq: 2 INVITE > Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, > MESSAGE > Content-Type: application/sdp > Supported: replaces > User-Agent: X-Lite release 4.9.3 stamp 79961 > Content-Length: 333 > P-hint: NAT > > > BYE Opensips -> Customer > > *U 199.199.199.199:5060 <http://199.199.199.199:5060> -> 172.20.17.11:5060 > <http://172.20.17.11:5060>* > BYE sip:100111@200.200.200.200:27923;rinstance=b4a1be0f56d73cfd SIP/2.0 > Via: SIP/2.0/UDP 199.199.199.199:5060;branch=z9hG4bKe99a.e1c423a3.0 > To: <sip:100111@172.20.17.11>;tag=7db6f42e > From: <sip:551133333...@plat.test.com>;tag=as4088ffc9 > CSeq: 1 BYE > Call-ID: 79961NjFjMjY2MDhiYjhhNDYyN2UyMWZiMzFiNWM5NzBkNTM > Route: <sip:172.20.17.11;lr;ftag=7db6f42e;did=769.5ee54854> > Max-Forwards: 70 > Content-Length: 0 > User-Agent: Softswitch > > > On the initial invite, I fixed the Contact using fix_nated_contact() and > signalling works fine between A and B sides, the problem is happening when > Opensips hangup the call, because A side doesnt receive the BYE. > > Do you guys have an idea on how to fix this? Maybe is it a bug? > > Thanks > > > > > > _______________________________________________ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > >
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