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>
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 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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