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

[Bogdan-Andrei Iancu]

Hi Daniel,

There is no easy fix to come into my mind right now. But I will do more 
thinking on this.

BTW, please check as you have the same routing issue also when the B 
party sends the BYE to hung up -> that BYE also will get stuck on the way.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 23.04.2016 06:29, Daniel Zanutti wrote:
>
> 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" <bogdan at opensips.org 
> <mailto:bogdan at 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 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>
>>     <mailto: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
>>     <mailto: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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>

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