[OpenSIPS-Users] BYE generated locally going to private IP, not public IP

[Bogdan-Andrei Iancu]

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 Developer
http://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:551133333333 at plat.test.com 
> <mailto:sip%3A551133333333 at 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 at 172.28.0.12:57744;rinstance=b4a1be0f56d73cfd>
> To: <sip:551133333333 at plat.test.com 
> <mailto:sip%3A551133333333 at plat.test.com>>
> From: <sip:100111 at 172.20.17.11 
> <mailto:sip%3A100111 at 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 at 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 at 172.20.17.11 
> <mailto:sip%3A100111 at 172.20.17.11>>;tag=7db6f42e
> From: <sip:551133333333 at plat.test.com 
> <mailto:sip%3A551133333333 at 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
>
>
>
>
>
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