SIP2SIP

== Siemens Gigaset ==

||Connection Name or Number:|| SIP2SIP||

||Provider|| Other Provider||

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

||Authentication password|| Your sip2sip password||

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

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

||Domain || sip2sip.info||

||STUN enabled|| No||

||Outbound proxy mode|| Always||

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

=== Issues ===

Gigaset SIP devices have two major bugs:

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:

{{{

SENDING: Packet 69, +0:00:12.659560

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

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

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

Max-Forwards: 70

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

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

Contact: <sip:qjeszynx@192.168.1.6:63990>

Call-ID: eP2QcqdwqXg2uyeQctb2TsJ9Qcgvj3e8

CSeq: 26307 INVITE

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

Supported: 100rel, replaces, norefersub

User-Agent: Blink Pro 1.3.1 (MacOSX)

Content-Type: application/sdp

Content-Length:   305

v=0

o=- 3527679880 3527679880 IN IP4 192.168.1.6

s=Blink Pro 1.3.1 (MacOSX)

c=IN IP4 192.168.1.6

t=0 0

m=message 2855 TCP/TLS/MSRP *

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

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

a=accept-wrapped-types:*

a=setup:active

RECEIVED: Packet 70, +0:00:12.679434

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

SIP/2.0 100 Trying

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

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

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

Call-ID: eP2QcqdwqXg2uyeQctb2TsJ9Qcgvj3e8

CSeq: 26307 INVITE

Server: SIP Thor on OpenSIPS XS 1.4.5

Content-Length: 0

RECEIVED: Packet 76, +0:00:18.532525

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

SIP/2.0 200 OK

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

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

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

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

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

Call-ID: eP2QcqdwqXg2uyeQctb2TsJ9Qcgvj3e8

CSeq: 26307 INVITE

Contact: <sip:31208005169@95.97.50.27:61000>

Supported: replaces

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

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

Content-Type: application/sdp

Content-Length: 206

v=0

o=31208005169 5020 12 IN IP4 192.168.1.131

s=Mapping

c=IN IP4 81.23.228.150

t=0 0

m=audio 56656 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

SENDING: Packet 77, +0:00:18.532887

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

ACK sip:31208005169@95.97.50.27:61000 SIP/2.0

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

Max-Forwards: 70

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

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

Call-ID: eP2QcqdwqXg2uyeQctb2TsJ9Qcgvj3e8

CSeq: 26307 ACK

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

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

User-Agent: Blink Pro 1.3.1 (MacOSX)

Content-Length:  0

}}}

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.