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.