[Freeswitch-users] FreeSWITCH using same Call-ID for forked calls
Brian West
brian at freeswitch.org
Wed Mar 18 21:57:25 MSK 2015
You're using 1.2, I see 1.2.12 and 1.2.7 in your user agents above, I would
highly recommend you re-test with Master or at the very least 1.4.17 or
1.4.18 which should be out later today.
1.2 is not receiving patches, updates or support moving forward, our
release branch is 1.4.x
On Wed, Mar 18, 2015 at 1:26 PM, Örn Arnarson <orn at arnarson.net> wrote:
> Hello,
>
> Not sure whether this belong in the users list or the dev list, but when
> in doubt; start with users :-)
>
> I am using FreeSWITCH as an SBC, talking to Kamailio on one and and
> Asterisk on the other, and am seeing some strange behavior when calls are
> being forked on the Asterisk.
>
> Call setup is like this:
> 1. FreeSWITCH receives INVITE from Kamailio
> 2. FreeSWITCH sends INVITE to Asterisk with new Call-ID
> 3. Asterisk forks call, sends out multiple INVITEs back to FreeSWITCH
> (each with a unique call-id)
> 4. FreeSWITCH sends multiple INVITEs to Kamailio, each with the new
> Call-ID from step 2.
>
> This is causing problems with one of the MGWs behind Kamailio, which is
> seeing multiple INVITEs to different destinations with the same Call-ID.
>
> So, firstly, why is FreeSWITCH reusing call-ids?
>
> Secondly, how is it matching up the calls? I can't find anything common in
> the INVITEs, other than the source number and obviously that the IP sent to
> and received from is the same.
>
> I'm not sure if this is intended behavior or not, but is there a way to
> have FreeSWITCH not do that?
>
> Regards,
> Örn
>
> P.S. Here is the sequence of INVITEs. I also have the console log (for a
> different call) if needed.
>
> *INVITE sent to FreeSWITCH by Kamailio:*
> INVITE sip:5344446 at 172.25.200.111:5080 SIP/2.0
> Record-Route: <sip:172.25.200.101;lr=on>
> Via: SIP/2.0/UDP 172.25.200.101;branch=z9hG4bK165.08b3b5c4.0
> Via: SIP/2.0/UDP 172.25.200.121:5080
> ;rport=5080;branch=z9hG4bK9aFD5m2KKerHN
> Max-Forwards: 16
> From: "4151502" <sip:4151502 at 172.25.200.121>;tag=33vB4BmmDtU0B
> To: <sip:5344446 at 172.25.200.101>
> Call-ID: 84b63791-4839-1233-639f-00215e2db0e0
> CSeq: 73014324 INVITE
> Contact: <sip:mod_sofia at 172.25.200.121:5080>
> User-Agent: FreeSWITCH-mod_sofia/1.2.7
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, conference, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 229
> X-FS-Support: update_display,send_info
> Remote-Party-ID: "4151502" <sip:4151502 at 172.25.200.121
> >;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1426681750 1426681751 IN IP4 172.25.200.121
> s=FreeSWITCH
> c=IN IP4 172.25.200.121
> t=0 0
> m=audio 19026 RTP/AVP 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
>
> *INVITE sent to Asterisk by FreeSWITCH:*
> INVITE sip:5344446 at 172.26.0.62:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.11.12.13;rport;branch=z9hG4bK0N733e7FHv4QF
> Max-Forwards: 15
> From: "4151502" <sip:4151502 at 10.11.12.13>;tag=2BaZj0t076Q9B
> To: <sip:5344446 at 172.26.0.62:5060>
> Call-ID: a7c77ea5-4839-1233-73b9-00215e2c8c90
> CSeq: 73014353 INVITE
> Contact: <sip:mod_sofia at 10.11.12.13:5060>
> User-Agent: FreeSWITCH-mod_sofia/1.2.12
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, conference, presence, dialog, line-seize,
> call-info, sla, include-session-description, presence.winfo,
> message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 223
> X-FS-Support: update_display,send_info
> Remote-Party-ID: "4151502" <sip:4151502 at 10.11.12.13
> >;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1426677605 1426677606 IN IP4 10.11.12.13
> s=FreeSWITCH
> c=IN IP4 10.11.12.13
> t=0 0
> m=audio 23230 RTP/AVP 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
> *First INVITE sent to FreeSWITCH by Asterisk (forked call):*
> INVITE sip:7712552 at 10.11.12.13 SIP/2.0
> Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK416db3f1;rport
> Max-Forwards: 70
> From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as24a51ba6
> To: <sip:7712552 at 10.11.12.13>
> Contact: <sip:4151502 at 172.26.0.62:5060>
> Call-ID: 135674a534fad0fd5bfff55c2fdc3280 at 172.26.0.62:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.8.15-cert2
> Date: Wed, 18 Mar 2015 17:47:11 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces, timer
> Diversion: <sip:5344446 at 172.26.0.62>
> Content-Type: application/sdp
> Content-Length: 312
>
> v=0
> o=root 693576967 693576967 IN IP4 172.26.0.62
> s=Asterisk PBX 1.8.15-cert2
> c=IN IP4 172.26.0.62
> t=0 0
> m=audio 30440 RTP/AVP 8 0 9 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>
> Second INVITE sent to FreeSWITCH by Asterisk (forked call):
> INVITE sip:6595454 at 10.11.12.13 SIP/2.0
> Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK6796aff1;rport
> Max-Forwards: 70
> From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as22f810b0
> To: <sip:6595454 at 10.11.12.13>
> Contact: <sip:4151502 at 172.26.0.62:5060>
> Call-ID: 6979c3dd69c5f8e557131e485466ad57 at 172.26.0.62:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.8.15-cert2
> Date: Wed, 18 Mar 2015 17:47:11 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces, timer
> Diversion: <sip:5344446 at 172.26.0.62>
> Content-Type: application/sdp
> Content-Length: 310
>
> v=0
> o=root 89056081 89056081 IN IP4 172.26.0.62
> s=Asterisk PBX 1.8.15-cert2
> c=IN IP4 172.26.0.62
> t=0 0
> m=audio 30708 RTP/AVP 8 0 9 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> *First INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):*
> INVITE sip:7712552 at 172.25.200.101 SIP/2.0
> Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK3jvS9UyXU216H
> Max-Forwards: 69
> From: "4151502" <sip:4151502 at 172.25.200.111>;tag=Z6pSHe2eXSB2p
> To: <sip:7712552 at 172.25.200.101>
> Call-ID: a7d2b58b-4839-1233-73b9-00215e2c8c90
> CSeq: 73014353 INVITE
> Contact: <sip:mod_sofia at 172.25.200.111:5080>
> User-Agent: FreeSWITCH-mod_sofia/1.2.12
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, conference, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 209
> Diversion: <sip:5344446 at 172.25.200.101>
> X-FS-Support: update_display,send_info
>
> v=0
> o=FreeSWITCH 1426681031 1426681032 IN IP4 172.25.200.111
> s=FreeSWITCH
> c=IN IP4 172.25.200.111
> t=0 0
> m=audio 19804 RTP/AVP 8 0 9 101 13
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> *Second INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):*
> INVITE sip:6595454 at 172.25.200.101 SIP/2.0
> Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK4UNjBQF1rBrSD
> Max-Forwards: 69
> From: "4151502" <sip:4151502 at 172.25.200.111>;tag=0FgjK9jjt21mj
> To: <sip:6595454 at 172.25.200.101>
> Call-ID: a7d2dd62-4839-1233-73b9-00215e2c8c90
> CSeq: 73014353 INVITE
> Contact: <sip:mod_sofia at 172.25.200.111:5080>
> User-Agent: FreeSWITCH-mod_sofia/1.2.12
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, conference, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 209
> Diversion: <sip:5344446 at 172.25.200.101>
> X-FS-Support: update_display,send_info
>
> v=0
> o=FreeSWITCH 1426669459 1426669460 IN IP4 172.25.200.111
> s=FreeSWITCH
> c=IN IP4 172.25.200.111
> t=0 0
> m=audio 31376 RTP/AVP 8 0 9 101 13
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
>
> _________________________________________________________________________
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--
*Brian West*
brian at freeswitch.org
*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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