[Freeswitch-users] SIP 100 response and media

Thu Jul 7 17:06:07 MSD 2011

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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-- 
Kristian Kielhofner