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" <bog...@opensips.org
<mailto: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 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:551133333...@plat.test.com
<mailto:sip%3a551133333...@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>
<mailto:sip:100111@172.28.0.12:57744;rinstance=b4a1be0f56d73cfd>
To: <sip:551133333...@plat.test.com
<mailto:sip%3a551133333...@plat.test.com>>
From: <sip:100111@172.20.17.11
<mailto:sip%3A100111@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
<mailto: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
<mailto:sip%3A100111@172.20.17.11>>;tag=7db6f42e
From: <sip:551133333...@plat.test.com
<mailto:sip%3a551133333...@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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