FS Confusion with multiple SIP Record-Route headers?
Hello everyone, I have an issue where FS seems to get confused in the presence of multiple Record-Route headers. SIP capture here: http://admin.star2star.com/fs-sip.log I've never seen this with FS before but it appears to process the multiple Record-Route headers backwards, at least in this case. I want to verify: 1) These Record-Route headers are syntactically correct (looks good to me). 2) FS should, in fact, process Record-Route headers "top down" and built its Route: headers (and reply) accordingly. At first I thought the FS/Sofia SIP parser may have been getting confused because the Record-Route from my proxy (.186) does not have a port in the URI. I tried adding a Record-Route header with a port - no difference. This is currently running trunk rev 12218 but I'm about to update to 12571 to see what happens. Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com
On Wed, Mar 11, 2009 at 2:54 PM, Michael Collins <msc@freeswitch.org> wrote:
This is currently running trunk rev 12218 but I'm about to update to 12571 to see what happens.
To quote Samuel L. Jackson in "Jurassic Park": Hold on to your butts!
-MC
Yeah, I know. It's just that it's an AstLinux machine and my build machine is REALLY slow... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com
On Wed, Mar 11, 2009 at 2:54 PM, Michael Collins <msc@freeswitch.org> wrote:
This is currently running trunk rev 12218 but I'm about to update to 12571 to see what happens.
To quote Samuel L. Jackson in "Jurassic Park": Hold on to your butts!
-MC
Just got around to trying again on 12571 - same result. Here it is again just the OK and the ACK this time: U 208.38.149.186:5060 -> 71.228.78.51:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 71.228.78.51:5060;received=71.228.78.51;rport=5060;branch=z9hG4bK4D194FH359yaH. To: <sip:19412848354@208.38.149.186>;tag=as4ba5ae74. From: "Extension 1000" <sip:1000@71.228.78.51>;tag=4veH7cg0XS04r. Call-ID: dec5a4c0-8951-122c-a78f-f96fa82849d1. CSeq: 112294452 INVITE. Content-Type: application/sdp. Contact: <sip:19412848354@64.2.142.73>. Content-Length: 285. Record-Route: <sip:208.38.149.186;lr;ftag=4veH7cg0XS04r>. User-Agent: Packetrino. Supported: replaces. Record-Route: <sip:64.2.142.93:5060;lr>. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. . v=0. o=root 10960 10961 IN IP4 64.2.142.73. s=session. c=IN IP4 64.2.142.73. t=0 0. m=audio 18680 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 71.228.78.51:5060 -> 64.2.142.93:5060 ACK sip:19412848354@64.2.142.73 SIP/2.0. Via: SIP/2.0/UDP 71.228.78.51;rport;branch=z9hG4bK5pt26a262jNXc. Route: <sip:64.2.142.93:5060;lr>. Route: <sip:208.38.149.186;lr;ftag=4veH7cg0XS04r>. Max-Forwards: 70. From: "Extension 1000" <sip:1000@71.228.78.51>;tag=4veH7cg0XS04r. To: <sip:19412848354@208.38.149.186>;tag=as4ba5ae74. Call-ID: dec5a4c0-8951-122c-a78f-f96fa82849d1. CSeq: 112294452 ACK. Contact: <sip:mod_sofia@71.228.78.51:5060>. Content-Length: 0. . Trying to be as self-sufficient as I can, it looks like the code for this is on line 4268 of src/mod/endpoints/mod_sofia/sofia.c. I just wish I new what to do to it... ;) Am I the only one that has experienced this? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com
After reading into this more it looks like the Record-Route headers are to be parsed in reverse order (which FS is doing). Sorry! On Wed, Mar 11, 2009 at 10:54 PM, Kristian Kielhofner <kristian.kielhofner@gmail.com> wrote:
On Wed, Mar 11, 2009 at 2:54 PM, Michael Collins <msc@freeswitch.org> wrote:
This is currently running trunk rev 12218 but I'm about to update to 12571 to see what happens.
To quote Samuel L. Jackson in "Jurassic Park": Hold on to your butts!
-MC
Just got around to trying again on 12571 - same result. Here it is again just the OK and the ACK this time:
U 208.38.149.186:5060 -> 71.228.78.51:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 71.228.78.51:5060;received=71.228.78.51;rport=5060;branch=z9hG4bK4D194FH359yaH. To: <sip:19412848354@208.38.149.186>;tag=as4ba5ae74. From: "Extension 1000" <sip:1000@71.228.78.51>;tag=4veH7cg0XS04r. Call-ID: dec5a4c0-8951-122c-a78f-f96fa82849d1. CSeq: 112294452 INVITE. Content-Type: application/sdp. Contact: <sip:19412848354@64.2.142.73>. Content-Length: 285. Record-Route: <sip:208.38.149.186;lr;ftag=4veH7cg0XS04r>. User-Agent: Packetrino. Supported: replaces. Record-Route: <sip:64.2.142.93:5060;lr>. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. . v=0. o=root 10960 10961 IN IP4 64.2.142.73. s=session. c=IN IP4 64.2.142.73. t=0 0. m=audio 18680 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv.
U 71.228.78.51:5060 -> 64.2.142.93:5060 ACK sip:19412848354@64.2.142.73 SIP/2.0. Via: SIP/2.0/UDP 71.228.78.51;rport;branch=z9hG4bK5pt26a262jNXc. Route: <sip:64.2.142.93:5060;lr>. Route: <sip:208.38.149.186;lr;ftag=4veH7cg0XS04r>. Max-Forwards: 70. From: "Extension 1000" <sip:1000@71.228.78.51>;tag=4veH7cg0XS04r. To: <sip:19412848354@208.38.149.186>;tag=as4ba5ae74. Call-ID: dec5a4c0-8951-122c-a78f-f96fa82849d1. CSeq: 112294452 ACK. Contact: <sip:mod_sofia@71.228.78.51:5060>. Content-Length: 0. .
Trying to be as self-sufficient as I can, it looks like the code for this is on line 4268 of src/mod/endpoints/mod_sofia/sofia.c. I just wish I new what to do to it... ;)
Am I the only one that has experienced this?
-- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com
-- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com
Aww, thanks Brian! On Wed, Mar 11, 2009 at 11:18 PM, Brian West <brian@freeswitch.org> wrote:
Its ok... at the very least you question the norm! ;)
/b
-- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com
On Mar 11, 2009, at 8:34 PM, Brian West <brian@freeswitch.org> wrote:
;) I expect to see you at cluecon this year?
/b
Notice how he threw in a compliment and an invite to CC but didn't actually address the question? ;) pretty sneaky bkw! BTW, the best way to come to CC is to get your boss to sponsor the event! :p -MC
I'm like bugs bunny here... sneaky wabbit! /b On Mar 11, 2009, at 10:50 PM, Michael S Collins wrote:
On Mar 11, 2009, at 8:34 PM, Brian West <brian@freeswitch.org> wrote:
;) I expect to see you at cluecon this year?
/b
Notice how he threw in a compliment and an invite to CC but didn't actually address the question? ;) pretty sneaky bkw!
BTW, the best way to come to CC is to get your boss to sponsor the event! :p
-MC
participants (4)
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Brian West
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Kristian Kielhofner
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Michael Collins
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Michael S Collins