[OpenSIPS-Users] Sending ACK from different port

Bogdan-Andrei Iancu bogdan at opensips.org
Tue Apr 16 11:19:39 CEST 2013


Hello Andrei,

OpenSIPS preserves the interface when doing a relay (uses as outbound 
same interface as inbound) if not otherwise instructed by routing info 
(Route hdrs) or scripting.

As ACK is a sequential request which is routed based on Route hdr, to 
see where the problem is, it is a must to see the full call capture from 
OpenSIPS - from first INVITE to ACK, both messages coming and leaving 
OpenSIPS - please post it on pastebin or so.

Best regards,

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


On 04/12/2013 11:10 PM, Andrei Grav wrote:
> Hi,
>
> I am facing some strange situation.
> Opensips is listening on multiple ports on a single public IP 
> 193.xx.xx.20 on ports: 5060, 26999, 36999
> Asterisk is on 193.xx.xx.24:5060
>
> Sometimes Opensips respond from 5060 to a 200OK instead received port.
>
>
> U 188.xx.xx.173:53929 -> 193.xx.xx.20:26999
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 
> 193.xx.xx.20:26999;received=193.xx.xx.20;branch=z9hG4bK46cd.df220482.1.
> Via: SIP/2.0/UDP 
> 193.xx.xx.24:5060;rport=5060;received=193.xx.xx.24;branch=z9hG4bK218bf88e.
> Record-Route: 
> <sip:193.xx.xx.20:26999;lr;r2=on;ftag=as61ef6194;did=49d.46da2342>.
> Record-Route: 
> <sip:193.xx.xx.20;lr;r2=on;ftag=as61ef6194;did=49d.46da2342>.
> Call-ID: 5ebb1c01580da34c694b320e0627dd7f at sip.mydomain.com 
> <mailto:5ebb1c01580da34c694b320e0627dd7f at sip.mydomain.com>.
> From: "User" <sip:850010 at sip.mydomain.com 
> <mailto:sip%3A850010 at sip.mydomain.com>>;tag=as61ef6194.
> To: <sip:850105 at sip.mydomain.com 
> <mailto:sip%3A850105 at sip.mydomain.com>>;tag=Yab-MAmXkF5lAQq2E6QpifQ8xa9xDoU7.
> CSeq: 102 INVITE.
> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, 
> REFER, MESSAGE, OPTIONS.
> Contact: <sip:850105 at 188.xx.xx.173:53929;ob>.
> Supported: replaces, 100rel, timer, norefersub.
> Content-Type: application/sdp.
> Content-Length: 294.
> .
> v=0.
> o=- 3574781465 3574781466 IN IP4 188.xx.xx.173.
> s=pjmedia.
> c=IN IP4 188.xx.xx.173.
> t=0 0.
> m=audio 4006 RTP/AVP 18 101.
> c=IN IP4 188.xx.xx.173.
> a=rtcp:4007 IN IP4 188.xx.xx.173.
> a=sendrecv.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
>
>
> U 193.xx.xx.20:5060 -> 188.xx.xx.173:53929
> ACK sip:850105 at 188.xx.xx.173:53929;ob SIP/2.0.
> Via: SIP/2.0/UDP 193.xx.xx.20:26999;branch=z9hG4bK46cd.df220482.3.
> Via: SIP/2.0/UDP 
> 193.xx.xx.24:5060;rport=5060;received=193.xx.xx.24;branch=z9hG4bK63fbcba6.
> Max-Forwards: 69.
> From: "User" <sip:850010 at sip.mydomain.com 
> <mailto:sip%3A850010 at sip.mydomain.com>>;tag=as61ef6194.
> To: <sip:850105 at sip.mydomain.com 
> <mailto:sip%3A850105 at sip.mydomain.com>>;tag=Yab-MAmXkF5lAQq2E6QpifQ8xa9xDoU7.
> Contact: <sip:850010 at 193.xx.xx.24:5060>.
> Call-ID: 5ebb1c01580da34c694b320e0627dd7f at sip.mydomain.com 
> <mailto:5ebb1c01580da34c694b320e0627dd7f at sip.mydomain.com>.
> CSeq: 102 ACK.
> User-Agent: PBX.
> Content-Length: 0.
> .
>
>
> the last response should send the response from port 26999 to be ok 
> ... or the call is hanged up after 32 seconds
> U 193.xx.xx.20:26999 -> 188.xx.xx.173:53929
>
> any advice ?
>
> Thank you,
> Andrei
>
>
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