[Freeswitch-users] SIP 100 response and media

Thu Jul 7 18:36:20 MSD 2011

Hello!

     I tried but.... it works only with <action application="export" 
data="verbose_sdp=true"/>, it this right? I thinked simple "set" is enough.
> http://wiki.freeswitch.org/wiki/Variable_verbose_sdp
>
> Set this to true in your dialplan before calling bridge when placing a
> call to the Cisco.
>
> On Thu, Jul 7, 2011 at 2:07 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>>      This thread again. There are two logs. First between Cisco<->Cisco
>> call and second between Cisco<->FreeSwitch. Cisco<->Cisco works fine. As
>> I can see the difference is that
>> Cisco has a=rtpmap:8 PCMA/8000 and FreeSwitch not. May be this is a
>> problem? Is this field mandatory?
>>
>>
>> Jul  6 18:20:39.341: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>> Received:
>> INVITE sip:28032 at 192.168.35.15:5060 SIP/2.0
>> Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F
>> Remote-Party-ID: "test"
>> <sip:30030 at 10.240.14.15>;party=calling;screen=no;privacy=off
>> From: "test"<sip:30030 at 10.240.14.15>;tag=3F5B1E64-2263
>> To:<sip:28032 at 192.168.35.15>
>> Date: Wed, 06 Jul 2011 18:20:39 GMT
>> Call-ID: 7C686709-A73311E0-95798064-7F05C5CA at 10.240.14.15
>> Supported: 100rel,timer,resource-priority,replaces,sdp-anat
>> Min-SE:  1800
>> Cisco-Guid: 2076058060-2805141984-2507440228-2131084746
>> User-Agent: Cisco-SIPGateway/IOS-12.x
>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>> CSeq: 101 INVITE
>> Max-Forwards: 70
>> Timestamp: 1309976439
>> Contact:<sip:30030 at 10.240.14.15:5060>
>> Expires: 180
>> Allow-Events: telephone-event
>> Content-Type: application/sdp
>> Content-Disposition: session;handling=required
>> Content-Length: 187
>>
>> v=0
>> o=CiscoSystemsSIP-GW-UserAgent 0 703 IN IP4 10.240.14.15
>> s=SIP Call
>> c=IN IP4 10.240.14.15
>> t=0 0
>> m=audio 18720 RTP/AVP 8
>> c=IN IP4 10.240.14.15
>> a=rtpmap:8 PCMA/8000
>> a=ptime:20
>>
>> Jul  6 18:20:39.361: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>> Sent:
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F
>> From: "test"<sip:30030 at 10.240.14.15>;tag=3F5B1E64-2263
>> To:<sip:28032 at 192.168.35.15>
>> Date: Wed, 06 Jul 2011 18:20:39 GMT
>> Call-ID: 7C686709-A73311E0-95798064-7F05C5CA at 10.240.14.15
>> Timestamp: 1309976439
>> CSeq: 101 INVITE
>> Allow-Events: telephone-event
>> Server: Cisco-SIPGateway/IOS-12.x
>> Content-Length: 0
>>
>>
>> Jul  6 18:20:39.369: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>> Sent:
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F
>> From: "test"<sip:30030 at 10.240.14.15>;tag=3F5B1E64-2263
>> To:<sip:28032 at 192.168.35.15>;tag=CBEFC0-16E4
>> Date: Wed, 06 Jul 2011 18:20:39 GMT
>> Call-ID: 7C686709-A73311E0-95798064-7F05C5CA at 10.240.14.15
>> Timestamp: 1309976439
>> CSeq: 101 INVITE
>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>> Allow-Events: telephone-event
>> Contact:<sip:28032 at 192.168.35.15:5060>
>> Supported: replaces
>> Supported: sdp-anat
>> Server: Cisco-SIPGateway/IOS-12.x
>> Content-Type: application/sdp
>> Content-Disposition: session;handling=required
>> Content-Length: 194
>>
>>
>> When call is between Cisco<->Freeswitch messages are
>> Jul  6 18:21:11.966: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>> Received:
>> INVITE sip:73435230030 at 192.168.3.6:5060 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp
>> Max-Forwards: 67
>> From: "anonymous"<sip:0000000000 at 192.168.1.1>;tag=QFKH52Z9NaN2c
>> To:<sip:73435230030 at 192.168.3.6:5060>
>> Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac
>> CSeq: 14652171 INVITE
>> Contact:<sip:mod_sofia at 192.168.1.1:5060>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1c95ad9 2011-01-20
>> 22-43-50 -0300
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE
>> Supported: 100rel, timer, precondition, path, replaces
>> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 329
>> X-FS-Support: update_display
>> Remote-Party-ID: "anonymous"
>> <sip:0000000000 at 192.168.1.1>;party=calling;screen=yes;privacy=off
>>
>>
>> v=0
>> o=FreeSWITCH 1309956111 1309956112 IN IP4 192.168.1.1
>> s=FreeSWITCH
>> c=IN IP4 192.168.1.1
>> t=0 0
>>
>> m=audio 20360 RTP/AVP 8 0 18 101 13
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> m=video 19030 RTP/AVP 31 34 98
>> a=rtpmap:31 H261/90000
>> a=rtpmap:34 H263/90000
>> a=rtpmap:98 H264/90000
>>
>>
>> Jul  6 18:21:11.986: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>> Sent:
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp
>> From: "anonymous"<sip:0000000000 at 192.168.1.1>;tag=QFKH52Z9NaN2c
>> To:<sip:73435230030 at 192.168.3.6:5060>
>> Date: Wed, 06 Jul 2011 18:21:11 GMT
>> Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac
>> CSeq: 14652171 INVITE
>> Allow-Events: telephone-event
>> Server: Cisco-SIPGateway/IOS-12.x
>> Content-Length: 0
>>
>>
>>
>>
>>
>> --
>> С уважением,
>>    Борис Коваленко
>>    ЗАО "Тагнет"
>>    тел. +7 (3435) 230001
>>    факс +7 (3435) 230005
>>
>>
>> _______________________________________________
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>>
>
>

-- 
С уважением,
   Борис Коваленко
   ЗАО "Тагнет"
   тел. +7 (3435) 230001
   факс +7 (3435) 230005