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SipDevicesGigaset » History » Version 3

Adrian Georgescu, 10/15/2011 05:07 PM

1 2 Adrian Georgescu 
== Siemens Gigaset ==
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||Connection Name or Number:|| SIP2SIP||





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||Provider|| Other Provider||





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||Authentication name|| Your sip2sip username without the domain part||





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||Authentication password|| Your sip2sip password||





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||Username|| Your sip2sip username without the domain part||





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||Display name || Your real name displayed on remote devices e.g. John Doe ||





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||Domain || sip2sip.info||





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||STUN enabled|| No||





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||Outbound proxy mode|| Always||





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||Outbound server address|| proxy.sipthor.net||





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    3
    Adrian Georgescu
    





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=== Issues ===





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Gigaset SIP devices have two major bugs:





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1. Incoming SIP sessions with no audio in SDP cause the phone to ring.  When user answers the media plane is broken as the phone does not support non-audio streams. For example this is an incoming text chat session (m=message in SDP) which is not supported by Gigaset. Instead of rejecting the session with 488 No compatible media or similar code, the phone answers the call with 200 OK and m=audio in SDP:





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{{{





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SENDING: Packet 69, +0:00:12.659560





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2011-10-15 17:04:40.767720: 192.168.1.163:63990 -(SIP over UDP)-> 85.17.186.7:5060





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INVITE sip:31208005169@ag-projects.com SIP/2.0





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Via: SIP/2.0/UDP 192.168.1.163:63990;rport;branch=z9hG4bKPjMMtvbefmIyvyBUzkCuWjTTVbR.leasGo





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Max-Forwards: 70





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From: "Adrian Georgescu" <sip:31208005169@ag-projects.com>;tag=c9gx4YtaXAGm8VozQCGvo9gDRRNUt69U





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To: <sip:31208005169@ag-projects.com>





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Contact: <sip:qjeszynx@192.168.1.6:63990>





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Call-ID: eP2QcqdwqXg2uyeQctb2TsJ9Qcgvj3e8





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CSeq: 26307 INVITE





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Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER





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Supported: 100rel, replaces, norefersub





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User-Agent: Blink Pro 1.3.1 (MacOSX)





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Content-Type: application/sdp





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Content-Length:   305





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v=0





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o=- 3527679880 3527679880 IN IP4 192.168.1.6





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s=Blink Pro 1.3.1 (MacOSX)





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c=IN IP4 192.168.1.6





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t=0 0





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m=message 2855 TCP/TLS/MSRP *





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a=path:msrps://192.168.1.6:2855/6d06f0d3b4a7a29a44f;tcp





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a=accept-types:message/cpim text/* application/im-iscomposing+xml





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a=accept-wrapped-types:*





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a=setup:active





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RECEIVED: Packet 70, +0:00:12.679434





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2011-10-15 17:04:40.787594: 85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.163:63990





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SIP/2.0 100 Trying





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Via: SIP/2.0/UDP 192.168.1.163:63990;rport=63990;branch=z9hG4bKPjMMtvbefmIyvyBUzkCuWjTTVbR.leasGo;received=95.97.50.27





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From: "Adrian Georgescu" <sip:31208005169@ag-projects.com>;tag=c9gx4YtaXAGm8VozQCGvo9gDRRNUt69U





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To: <sip:31208005169@ag-projects.com>





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Call-ID: eP2QcqdwqXg2uyeQctb2TsJ9Qcgvj3e8





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CSeq: 26307 INVITE





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Server: SIP Thor on OpenSIPS XS 1.4.5





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Content-Length: 0





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RECEIVED: Packet 76, +0:00:18.532525





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2011-10-15 17:04:46.640685: 85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.163:63990





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SIP/2.0 200 OK





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Via: SIP/2.0/UDP 192.168.1.163:63990;received=95.97.50.27;rport=63990;branch=z9hG4bKPjMMtvbefmIyvyBUzkCuWjTTVbR.leasGo





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Record-Route: <sip:81.23.228.129;lr;ftag=c9gx4YtaXAGm8VozQCGvo9gDRRNUt69U;did=3b2.e29756f7>





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Record-Route: <sip:85.17.186.7;lr;ftag=c9gx4YtaXAGm8VozQCGvo9gDRRNUt69U;did=3b2.1bef3ff>





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From: "Adrian Georgescu" <sip:31208005169@ag-projects.com>;tag=c9gx4YtaXAGm8VozQCGvo9gDRRNUt69U





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To: <sip:31208005169@ag-projects.com>;tag=1360047391





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Call-ID: eP2QcqdwqXg2uyeQctb2TsJ9Qcgvj3e8





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CSeq: 26307 INVITE





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Contact: <sip:31208005169@95.97.50.27:61000>





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Supported: replaces





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Allow-Events: message-summary, refer, ua-profile





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Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY





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Content-Type: application/sdp





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Content-Length: 206





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v=0





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o=31208005169 5020 12 IN IP4 192.168.1.131





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s=Mapping





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c=IN IP4 81.23.228.150





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t=0 0





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m=audio 56656 RTP/AVP 0 101





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a=rtpmap:0 PCMU/8000





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a=rtpmap:101 telephone-event/8000





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a=fmtp:101 0-16





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a=sendrecv





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SENDING: Packet 77, +0:00:18.532887





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2011-10-15 17:04:46.641047: 192.168.1.163:63990 -(SIP over UDP)-> 85.17.186.7:5060





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ACK sip:31208005169@95.97.50.27:61000 SIP/2.0





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Via: SIP/2.0/UDP 192.168.1.163:63990;rport;branch=z9hG4bKPj0mAfcWZh3Uq8yiEHO36mVtFI-A-v10mh





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Max-Forwards: 70





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From: "Adrian Georgescu" <sip:31208005169@ag-projects.com>;tag=c9gx4YtaXAGm8VozQCGvo9gDRRNUt69U





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To: <sip:31208005169@ag-projects.com>;tag=1360047391





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Call-ID: eP2QcqdwqXg2uyeQctb2TsJ9Qcgvj3e8





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CSeq: 26307 ACK





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Route: <sip:85.17.186.7;lr;ftag=c9gx4YtaXAGm8VozQCGvo9gDRRNUt69U;did=3b2.1bef3ff>





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Route: <sip:81.23.228.129;lr;ftag=c9gx4YtaXAGm8VozQCGvo9gDRRNUt69U;did=3b2.e29756f7>





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User-Agent: Blink Pro 1.3.1 (MacOSX)





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Content-Length:  0





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}}}





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2. Incoming SIP sessions without a phone number in the username part are displayed as Unknown. So a SIP call from a soft phone or any SIP device that has a non numerical SIP address in the From header confuses the phone0.