I'm having a problem using loose_route, and I'm not sure if it's a bug/feature/misunderstanding.
I have a B2BUA (Asterisk) running on the same system as OpenSER, and in my failure_route for voicemail I direct calls there which actually sends them on to the voicemail server. If someone 'zeroes out' of voicemail, to be forwarded to the operator, the voicemail server issues a re-INVITE to redirect media. This is where the problem comes in.
If I try to loose_route this re-INVITE, it fails. However, putting the B2BUA on a different system works fine.
This is running OpenSER 1.2.3 currently, though I'm not particularly averse to upgrading to the latest 1.3.x if for some reason this is something that's been resolved there.
Any suggestions?
Thanks in advance, - Brad
On Mon, Jun 9, 2008 at 6:35 PM, Watkins, Bradley Bradley.Watkins@compuware.com wrote:
I'm having a problem using loose_route, and I'm not sure if it's a bug/feature/misunderstanding.
I have a B2BUA (Asterisk) running on the same system as OpenSER, and in my failure_route for voicemail I direct calls there which actually sends them on to the voicemail server. If someone 'zeroes out' of voicemail, to be forwarded to the operator, the voicemail server issues a re-INVITE to redirect media. This is where the problem comes in.
If I try to loose_route this re-INVITE, it fails. However, putting the B2BUA on a different system works fine.
I'm not sure that this is a loose routing problem. A SIP message trace and an Asterisk 'sip set debug' trace can help identify the issue.
-- Raj Jain
-----Original Message----- From: Raj Jain [mailto:rj2807@gmail.com] Sent: Monday, June 09, 2008 6:46 PM To: Watkins, Bradley Cc: users@lists.openser.org Subject: Re: [OpenSER-Users] Loose route problem or misunderstanding
On Mon, Jun 9, 2008 at 6:35 PM, Watkins, Bradley Bradley.Watkins@compuware.com wrote:
I'm having a problem using loose_route, and I'm not sure if it's a bug/feature/misunderstanding.
I have a B2BUA (Asterisk) running on the same system as
OpenSER, and in
my failure_route for voicemail I direct calls there which
actually sends
them on to the voicemail server. If someone 'zeroes out'
of voicemail,
to be forwarded to the operator, the voicemail server
issues a re-INVITE
to redirect media. This is where the problem comes in.
If I try to loose_route this re-INVITE, it fails. However,
putting the
B2BUA on a different system works fine.
I'm not sure that this is a loose routing problem. A SIP message trace and an Asterisk 'sip set debug' trace can help identify the issue.
-- Raj Jain
I'm quite sure that it is, actually. Mind you, it could be that I'm expecting loose_route to do something that by RFC compliance it shouldn't, hence the admission that I may be misunderstanding.
Here's the relevant SIP messages for a failing scenario:
U 10.3.104.104:5060 -> 10.0.12.51:5060 INVITE sip:71841@10.0.12.51:5062 SIP/2.0. From: sip:77740@10.0.12.51;tag=6868030a-13c4-4848318e-d46a11b5-7671. To: "Brad Watkins"sip:71841@10.0.12.51:5062;tag=as75bb16a1. Call-ID: 75bd11cb40e6f85239b3bf6c525bb206@10.0.12.51. CSeq: 2 INVITE. Via: SIP/2.0/UDP 10.3.104.104:5060;branch=z9hG4bK-484831a0-d46a55df-21eb. Supported: 100rel,sipvc,replaces,timer. User-Agent: Nortel CS1000 SIP GW release_4.5 version_sse-4.50.88. P-Asserted-Identity: sip:5077740;phone-context=udp@compuware.com;user=phone. Privacy: none. Contact: sip:5077740;phone-context=udp@compuware.com:5060;maddr=10.3.104.104;tra nsport=udp;user=phone. Route: sip:10.0.12.51;lr;ftag=as75bb16a1. Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE ,UPDATE. Content-Type: application/SDP. Content-Length: 148. . v=0. o=- 121469 3 IN IP4 10.3.104.104. s=-. t=0 0. m=audio 5202 RTP/AVP 0 8 18. c=IN IP4 10.3.102.105. a=fmtp:18 annexb=no. a=ptime:20. a=sendrecv.
# U 10.0.12.51:5060 -> 10.3.104.104:5060 SIP/2.0 100 Giving a try. From: sip:77740@10.0.12.51;tag=6868030a-13c4-4848318e-d46a11b5-7671. To: "Brad Watkins"sip:71841@10.0.12.51:5062;tag=as75bb16a1. Call-ID: 75bd11cb40e6f85239b3bf6c525bb206@10.0.12.51. CSeq: 2 INVITE. Via: SIP/2.0/UDP 10.3.104.104:5060;branch=z9hG4bK-484831a0-d46a55df-21eb. Server: OpenSER (1.2.3-notls (i386/linux)). Content-Length: 0. .
# U 10.0.12.51:5060 -> 10.3.104.104:5060 SIP/2.0 404 Not here. From: sip:77740@10.0.12.51;tag=6868030a-13c4-4848318e-d46a11b5-7671. To: "Brad Watkins"sip:71841@10.0.12.51:5062;tag=as75bb16a1. Call-ID: 75bd11cb40e6f85239b3bf6c525bb206@10.0.12.51. CSeq: 2 INVITE. Via: SIP/2.0/UDP 10.3.104.104:5060;branch=z9hG4bK-484831a0-d46a55df-21eb. Server: OpenSER (1.2.3-notls (i386/linux)). Content-Length: 0.
This is with a basic loose-routing construct from the sample config:
if (has_totag()) { # sequential request within a dialog should # take the path determined by record-routing if (loose_route()) { loose_route(); route(1); } else { sl_send_reply("404","Not here"); } exit;
And here's a working snippet, with the exact same OpenSER config and Asterisk config (except bindport on Asterisk), except with Asterisk on a separate machine:
U 10.3.104.104:5060 -> 10.0.12.51:5060 INVITE sip:71841@10.0.12.52 SIP/2.0. From: sip:77740@10.0.12.51;tag=6868030a-13c4-484eb0f3-edcbf878-e5e. To: "Brad Watkins"sip:71841@10.0.12.52;tag=as7c7f08f2. Call-ID: 2058be9c5b530f237378e77f2c12f5dd@10.0.12.52. CSeq: 2 INVITE. Via: SIP/2.0/UDP 10.3.104.104:5060;branch=z9hG4bK-484eb100-edcc2deb-b36. Supported: 100rel,sipvc,replaces,timer. User-Agent: Nortel CS1000 SIP GW release_4.5 version_sse-4.50.88. P-Asserted-Identity: sip:5077740;phone-context=udp@compuware.com;user=phone. Privacy: none. Contact: sip:5077740;phone-context=udp@compuware.com:5060;maddr=10.3.104.104;tra nsport=udp;user=phone. Route: sip:10.0.12.51;lr;ftag=as7c7f08f2. Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE ,UPDATE. Content-Type: application/SDP. Content-Length: 148. . v=0. o=- 126019 3 IN IP4 10.3.104.104. s=-. t=0 0. m=audio 5214 RTP/AVP 0 8 18. c=IN IP4 10.3.102.107. a=fmtp:18 annexb=no. a=ptime:20. a=sendrecv.
# U 10.0.12.51:5060 -> 10.3.104.104:5060 SIP/2.0 100 Giving a try. From: sip:77740@10.0.12.51;tag=6868030a-13c4-484eb0f3-edcbf878-e5e. To: "Brad Watkins"sip:71841@10.0.12.52;tag=as7c7f08f2. Call-ID: 2058be9c5b530f237378e77f2c12f5dd@10.0.12.52. CSeq: 2 INVITE. Via: SIP/2.0/UDP 10.3.104.104:5060;branch=z9hG4bK-484eb100-edcc2deb-b36. Server: OpenSER (1.2.3-notls (i386/linux)). Content-Length: 0. .
# U 10.0.12.51:5060 -> 10.3.104.104:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.3.104.104:5060;branch=z9hG4bK-484eb100-edcc2deb-b36. From: sip:77740@10.0.12.51;tag=6868030a-13c4-484eb0f3-edcbf878-e5e. To: "Brad Watkins"sip:71841@10.0.12.52;tag=as7c7f08f2. Call-ID: 2058be9c5b530f237378e77f2c12f5dd@10.0.12.52. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Contact: sip:71841@10.0.12.52. Content-Type: application/sdp. Content-Length: 176. . v=0. o=root 407 409 IN IP4 10.0.12.52. s=session. c=IN IP4 10.0.12.52. t=0 0. m=audio 18948 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv.
El Martes, 10 de Junio de 2008, Watkins, Bradley escribió:
I'm quite sure that it is, actually. Mind you, it could be that I'm expecting loose_route to do something that by RFC compliance it shouldn't, hence the admission that I may be misunderstanding.
Here's the relevant SIP messages for a failing scenario:
This is an in-dialog request, so WHY the host of the RURI is the OpenSer IP (10.0.12.51) instead of the UA "Contact" IP? Of course this is incorrect. The RURI host in any in-dialog request must be the remote-target (this is: the host in the received "Contact" from the other end point).
Nortel CS1000? I've bad experiences with a Nortel CS2000 but it *does* well loose-routing not as in your case.
U 10.3.104.104:5060 -> 10.0.12.51:5060 INVITE sip:71841@10.0.12.51:5062 SIP/2.0. From: sip:77740@10.0.12.51;tag=6868030a-13c4-4848318e-d46a11b5-7671. To: "Brad Watkins"sip:71841@10.0.12.51:5062;tag=as75bb16a1. Call-ID: 75bd11cb40e6f85239b3bf6c525bb206@10.0.12.51. CSeq: 2 INVITE. Via: SIP/2.0/UDP 10.3.104.104:5060;branch=z9hG4bK-484831a0-d46a55df-21eb. Supported: 100rel,sipvc,replaces,timer. User-Agent: Nortel CS1000 SIP GW release_4.5 version_sse-4.50.88. P-Asserted-Identity: sip:5077740;phone-context=udp@compuware.com;user=phone. Privacy: none. Contact: sip:5077740;phone-context=udp@compuware.com:5060;maddr=10.3.104.104;tra nsport=udp;user=phone. Route: sip:10.0.12.51;lr;ftag=as75bb16a1. Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE ,UPDATE. Content-Type: application/SDP. Content-Length: 148. . v=0. o=- 121469 3 IN IP4 10.3.104.104. s=-. t=0 0. m=audio 5202 RTP/AVP 0 8 18. c=IN IP4 10.3.102.105. a=fmtp:18 annexb=no. a=ptime:20. a=sendrecv.
# U 10.0.12.51:5060 -> 10.3.104.104:5060 SIP/2.0 100 Giving a try. From: sip:77740@10.0.12.51;tag=6868030a-13c4-4848318e-d46a11b5-7671. To: "Brad Watkins"sip:71841@10.0.12.51:5062;tag=as75bb16a1. Call-ID: 75bd11cb40e6f85239b3bf6c525bb206@10.0.12.51. CSeq: 2 INVITE. Via: SIP/2.0/UDP 10.3.104.104:5060;branch=z9hG4bK-484831a0-d46a55df-21eb. Server: OpenSER (1.2.3-notls (i386/linux)). Content-Length: 0. .
# U 10.0.12.51:5060 -> 10.3.104.104:5060 SIP/2.0 404 Not here. From: sip:77740@10.0.12.51;tag=6868030a-13c4-4848318e-d46a11b5-7671. To: "Brad Watkins"sip:71841@10.0.12.51:5062;tag=as75bb16a1. Call-ID: 75bd11cb40e6f85239b3bf6c525bb206@10.0.12.51. CSeq: 2 INVITE. Via: SIP/2.0/UDP 10.3.104.104:5060;branch=z9hG4bK-484831a0-d46a55df-21eb. Server: OpenSER (1.2.3-notls (i386/linux)). Content-Length: 0.
This is with a basic loose-routing construct from the sample config:
if (has_totag()) { # sequential request within a dialog should # take the path determined by record-routing if (loose_route()) { loose_route();
Take off the second "loose_route()", it has been done into the "if" stament.
route(1); } else { sl_send_reply("404","Not here"); } exit;
loose_route() examines the top "Route" header (10.0.12.51) and matches it agains OpenSer knows IP's and hostnames (and domains).
If it matches (and it does it) it takes off the "Route" header and send the request to the URI indicated in the RURI (if there is not more "Route" headers), but note that the request RURI is: INVITE sip:71841@10.0.12.51:5062 SIP/2.0
So it's 10.0.12.51: OpenSer IP !!!!! so OpenSer routes the request to itself !!! When the request arrives again to OpenSer (looped) it has "To" tag but not "Route" header so OpenSer replies with a correct "404: Not here". It's 100% correct.
The problem is the host of the RURI in the in-dialog request. Could you show how the INITIAL INVITE arrives to the Nortel (in case a UAC calls the Nortel), or the 200 OK arriving to Nortel (if Nortel initiates the call).
-----Original Message----- From: users-bounces@lists.openser.org [mailto:users-bounces@lists.openser.org] On Behalf Of Iñaki Baz Castillo Sent: Tuesday, June 10, 2008 5:14 PM To: users@lists.openser.org Subject: Re: [OpenSER-Users] Loose route problem or misunderstanding
El Martes, 10 de Junio de 2008, Watkins, Bradley escribió:
I'm quite sure that it is, actually. Mind you, it could be that I'm expecting loose_route to do something that by RFC compliance it shouldn't, hence the admission that I may be misunderstanding.
Here's the relevant SIP messages for a failing scenario:
This is an in-dialog request, so WHY the host of the RURI is the OpenSer IP (10.0.12.51) instead of the UA "Contact" IP? Of course this is incorrect. The RURI host in any in-dialog request must be the remote-target (this is: the host in the received "Contact" from the other end point).
The remote target is on the same machine, but on a different port. OpenSER is listening on 5060, Asterisk is listening on 5062.
Nortel CS1000? I've bad experiences with a Nortel CS2000 but it *does* well loose-routing not as in your case.
This is a CS1k, but the question is a vagary of my setup. Unfortunately, this is one problem I can't blame on the Nortel. ;)
Not to worry, there are plenty of others that I know for sure I can. :)
loose_route() examines the top "Route" header (10.0.12.51) and matches it agains OpenSer knows IP's and hostnames (and domains).
If it matches (and it does it) it takes off the "Route" header and send the request to the URI indicated in the RURI (if there is not more "Route" headers), but note that the request RURI is: INVITE sip:71841@10.0.12.51:5062 SIP/2.0
Yes, but if it were doing that it would work fine. Again, the endpoint here is on the same system, but a different port.
So it's 10.0.12.51: OpenSer IP !!!!! so OpenSer routes the request to itself !!! When the request arrives again to OpenSer (looped) it has "To" tag but not "Route" header so OpenSer replies with a correct "404: Not here". It's 100% correct.
But it shouldn't be sending it to itself at all. If it is, then it's ignoring the port in the RURI and that's certainly incorrect.
The problem is the host of the RURI in the in-dialog request. Could you show how the INITIAL INVITE arrives to the Nortel (in case a UAC calls the Nortel), or the 200 OK arriving to Nortel (if Nortel initiates the call).
I can, but it will have to wait until tomorrow I'm afraid. I'm away from those systems for the evening.
El Miércoles, 11 de Junio de 2008, Watkins, Bradley escribió:
This is an in-dialog request, so WHY the host of the RURI is the OpenSer IP (10.0.12.51) instead of the UA "Contact" IP? Of course this is incorrect. The RURI host in any in-dialog request must be the remote-target (this is: the host in the received "Contact" from the other end point).
The remote target is on the same machine, but on a different port. OpenSER is listening on 5060, Asterisk is listening on 5062.
You are completely right, sorry, I didn't realize on the different port.
So a SIP trace of what happens in that machine would be needed. Could you provide a complete SIP flow? it's neccesary to see also the flow between OpenSer and Asterisk.
-----Original Message----- From: users-bounces@lists.openser.org [mailto:users-bounces@lists.openser.org] On Behalf Of Iñaki Baz Castillo Sent: Tuesday, June 10, 2008 6:35 PM To: users@lists.openser.org Subject: Re: [OpenSER-Users] Loose route problem or misunderstanding
El Miércoles, 11 de Junio de 2008, Watkins, Bradley escribió:
The remote target is on the same machine, but on a
different port. OpenSER
is listening on 5060, Asterisk is listening on 5062.
You are completely right, sorry, I didn't realize on the different port.
So a SIP trace of what happens in that machine would be needed. Could you provide a complete SIP flow? it's neccesary to see also the flow between OpenSer and Asterisk.
Apparently, it pays to read the documentation...
In particular, the entry in the core cookbook for alias:
"It is necessary to include the port (the port value used in the "port=" or "listen=" defintions) in the alias definition otherwise the loose_route() function will not work as expected for local forwards"
This was exactly my problem.
I'm not entirely sure I agree with assuming that OpenSER is responsible for EVERY port, even when it's configure to only listen on one. But at any rate, changing a few lines to include the port number makes this work for me so I won't complain too loudly.
Thanks for taking the time to try and help me with this issue.
Regards, - Brad
-
Iñaki Baz Castillo -
Raj Jain -
Watkins, Bradley