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kristian.kielhofner at...
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 Posted: Wed Mar 11, 2009 1:57 pm Post subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Rou |
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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
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msc at freeswitch.org
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kristian.kielhofner at...
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kristian.kielhofner at...
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| Posted: Wed Mar 11, 2009 10:07 pm Post subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Rou |
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On Wed, Mar 11, 2009 at 2:54 PM, Michael Collins <msc@freeswitch.org> wrote:
| Quote: | | Quote: | 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
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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kristian.kielhofner at...
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| Posted: Wed Mar 11, 2009 10:22 pm Post subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Rou |
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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:
| Quote: | On Wed, Mar 11, 2009 at 2:54 PM, Michael Collins <msc@freeswitch.org> wrote:
| Quote: | | Quote: | 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
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org |
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brian at freeswitch.org
Guest
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| Posted: Wed Mar 11, 2009 10:25 pm Post subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Rou |
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Its ok... at the very least you question the norm!
/b
On Mar 11, 2009, at 10:13 PM, Kristian Kielhofner wrote:
| Quote: | After reading into this more it looks like the Record-Route headers
are to be parsed in reverse order (which FS is doing).
Sorry!
|
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Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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kristian.kielhofner at...
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brian at freeswitch.org
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| Posted: Wed Mar 11, 2009 10:45 pm Post subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Rou |
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I expect to see you at cluecon this year?
/b
On Mar 11, 2009, at 10:29 PM, Kristian Kielhofner wrote:
| Quote: | Aww, thanks Brian!
On Wed, Mar 11, 2009 at 11:18 PM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:
| Quote: | Its ok... at the very least you question the norm!
/b |
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msc at freeswitch.org
Guest
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| Posted: Wed Mar 11, 2009 10:56 pm Post subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Rou |
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On Mar 11, 2009, at 8:34 PM, Brian West <brian@freeswitch.org> wrote:
| Quote: | 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
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Freeswitch-users@lists.freeswitch.org
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brian at freeswitch.org
Guest
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| Posted: Wed Mar 11, 2009 11:00 pm Post subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Rou |
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I'm like bugs bunny here... sneaky wabbit!
/b
On Mar 11, 2009, at 10:50 PM, Michael S Collins wrote:
| Quote: | On Mar 11, 2009, at 8:34 PM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:
| Quote: | 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 |
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