[Freeswitch-users] FreeSWITCH using same Call-ID for forked calls
Örn Arnarson
orn at arnarson.net
Thu Mar 19 19:26:20 MSK 2015
Hi,
Thanks for your response.
I installed FS from git on another server, and it still shows exactly the
same behavior. See the two new INVITES for the forked call outbound from FS:
INVITE sip:7712552 at 192.168.10.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.101:5080;rport;branch=z9hG4bK67cUpBr27p25r
Max-Forwards: 69
From: "..rn" <sip:5460000 at 192.168.10.101>;tag=eU3yr5rNjcvDa
To: <sip:7712552 at 192.168.10.3>
Call-ID: 174ff657-48f7-1233-d586-080027f911ca
CSeq: 73055034 INVITE
Contact: <sip:mod_sofia at 192.168.10.101:5080>
User-Agent:
FreeSWITCH-mod_sofia/1.5.15b+git~20150318T193012Z~21d1e6fc4b~32bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 251
Diversion: <sip:4151500 at 172.25.200.101>
X-FS-Support: update_display,send_info
v=0
o=FreeSWITCH 1426761247 1426761248 IN IP4 192.168.10.101
s=FreeSWITCH
c=IN IP4 192.168.10.101
t=0 0
m=audio 20950 RTP/AVP 8 0 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
INVITE sip:6595454 at 192.168.10.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.101:5080;rport;branch=z9hG4bK5yK2mg7yaecKD
Max-Forwards: 69
From: "..rn" <sip:5460000 at 192.168.10.101>;tag=Dja6pa8HN35te
To: <sip:6595454 at 192.168.10.3>
Call-ID: 174f7870-48f7-1233-d586-080027f911ca
CSeq: 73055034 INVITE
Contact: <sip:mod_sofia at 192.168.10.101:5080>
User-Agent:
FreeSWITCH-mod_sofia/1.5.15b+git~20150318T193012Z~21d1e6fc4b~32bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 251
Diversion: <sip:4151500 at 172.25.200.101>
X-FS-Support: update_display,send_info
v=0
o=FreeSWITCH 1426750991 1426750992 IN IP4 192.168.10.101
s=FreeSWITCH
c=IN IP4 192.168.10.101
t=0 0
m=audio 31206 RTP/AVP 8 0 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Regards,
Örn
On Wed, Mar 18, 2015 at 6:57 PM, Brian West <brian at freeswitch.org> wrote:
> 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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>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
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>>
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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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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