Nokia E71 SIP

wurstsalat

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Hallo zusammen,

ich habe ein Problem mein E71 via SIP an einem ePhone SIP-Gateway zum laufen zu bekommen.

Ausgangssituation:
- ePhone SIP-Gateway
- ePhone SIP-Softphone
- Polycom SIP-Hardphone
- Nokia E71 Symbian S60

Die ePhone Softphones können ohne Probleme untereinander telefonieren.
Ebenso wie ePhone zu Polycom Hardphone ohne Probleme in beide Richtungen funktioniert.
Nokia und Polycom können ebenfalls in beide Richtungen problemlos telefonieren.

Lediglich die Verbindung Nokia zu Softphone und umgekehrt funktioniert nicht :-/

Das Nokia registriert sich ordnungsgemäss, das Routing funktioniert jeweils, sprich es klingelt, aber sobald einer der Teilnehmer (Nokia oder Softphone) das Gespräch annehmen will wird sofort die Verbindung getrennt.

Der Trace bei einer funktionierenden Verbindung mit dem Polycom Hardphone zum Softphone sieht so aus:

Code:
SIPFlowMessage 1 - captured: 06/03/2009 17:26:21.665 CEST
INVITE sip:[email protected]:5080;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.28.186.50:5060;branch=z9hG4bK-c18e39de4df764390d559dca55c917ed
Via: SIP/2.0/UDP 10.28.175.175:5060;branch=z9hG4bK6178670eA3EEE34F
From: "SIP1079"<sip:[email protected]>;tag=EB16BBFB-FD9EDAC0
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Max-Forwards: 69
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Allow-Events: conference, hold, talk
Contact: <sip:[email protected]:5060>
Supported: 100rel, replaces
User-Agent: PolycomSoundStationIP-SSIP_4000-UA/3.1.2.0392
Content-Length: 223
Content-Type: application/sdp

v=0
o=- 1244016799 1244016799 IN IP4 10.28.175.175
s=Polycom IP Phone
c=IN IP4 10.28.175.175
t=0 0
a=sendrecv
m=audio 2224 RTP/AVP 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

Source: 10.28.186.50:5060
Destination: 10.28.175.64:5080

SIPFlowMessage 2 - captured: 06/03/2009 17:26:22.001 CEST
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.28.186.50:5060;branch=z9hG4bK-c18e39de4df764390d559dca55c917ed
Via: SIP/2.0/UDP 10.28.175.175:5060;branch=z9hG4bK6178670eA3EEE34F
From: "SIP1079"<sip:[email protected]>;tag=EB16BBFB-FD9EDAC0
To: <sip:[email protected];user=phone>;tag=160c0f00
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: "Username"<sip:[email protected]:5080;user=phone>
Content-Length: 0


Source: 10.28.175.64:5080
Destination: 10.28.186.50:5060

SIPFlowMessage 3 - captured: 06/03/2009 17:26:27.050 CEST
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.28.186.50:5060;branch=z9hG4bK-c18e39de4df764390d559dca55c917ed
Via: SIP/2.0/UDP 10.28.175.175:5060;branch=z9hG4bK6178670eA3EEE34F
From: "SIP1079"<sip:[email protected]>;tag=EB16BBFB-FD9EDAC0
To: <sip:[email protected];user=phone>;tag=160c0f00
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: "Username"<sip:[email protected]:5080;user=phone>
Content-Length: 207
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, TRANSFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Type: application/sdp

v=0
o=- 3453031586 3453031586 IN IP4 10.28.175.64
s=e-phone session
c=IN IP4 10.28.175.64
t=3453031586 0
m=audio 6000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20

Source: 10.28.175.64:5080
Destination: 10.28.186.50:5060

SIPFlowMessage 4 - captured: 06/03/2009 17:26:27.097 CEST
ACK sip:[email protected]:5080;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.28.175.175:5060;branch=z9hG4bK8ba460e9EC0F6246
From: "SIP1079" <sip:[email protected]>;tag=EB16BBFB-FD9EDAC0
To: <sip:[email protected];user=phone>;tag=160c0f00
CSeq: 1 ACK
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundStationIP-SSIP_4000-UA/3.1.2.0392
Accept-Language: en
Max-Forwards: 70
Content-Length: 0


Source: 10.28.175.175:5060
Destination: 10.28.175.64:5080

SIPFlowMessage 5 - captured: 06/03/2009 17:26:30.205 CEST
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.28.175.64:5080;branch=z9hG4bK-71416e920e203cb588b1b48efb5575ef
From: <sip:[email protected];user=phone>;tag=160c0f00
To: "SIP1079"<sip:[email protected]>;tag=EB16BBFB-FD9EDAC0
Call-ID: [email protected]
CSeq: 29018 BYE
Max-Forwards: 70
Content-Length: 0


Source: 10.28.175.64:5080
Destination: 10.28.175.175:5060

SIPFlowMessage 6 - captured: 06/03/2009 17:26:30.227 CEST
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.28.175.64:5080;branch=z9hG4bK-71416e920e203cb588b1b48efb5575ef
From: <sip:[email protected];user=phone>;tag=160c0f00
To: "SIP1079" <sip:[email protected]>;tag=EB16BBFB-FD9EDAC0
CSeq: 29018 BYE
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>
User-Agent: PolycomSoundStationIP-SSIP_4000-UA/3.1.2.0392
Accept-Language: en
Content-Length: 0


Source: 10.28.175.175:5060
Destination: 10.28.175.64:5080

Wenn ich mit dem Nokia versuche eine Verbindung zum Softphone aufzubauen sieht das so aus. Es klingelt, und sobald abgenommen wird bricht die Verbindung ab:

Code:
SIPFlowMessage 1 - captured: 06/03/2009 17:37:51.309 CEST
INVITE sip:[email protected]:5080;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.28.186.50:5060;branch=z9hG4bK-076e792d779e21aeea4edc3df0892a05
Via: SIP/2.0/UDP 10.28.185.239:5060;branch=z9hG4bK37r32rgn6dhc6cs3no8iq8n
From: <sip:1075@Netphoneserver>;tag=jtqj2rhqc9hc7d3ano8k
To: <sip:1430@Netphoneserver;user=phone>
Call-ID: FXINrGd5oIccbHUhL3BAaa0uo5ellK
CSeq: 955 INVITE
Max-Forwards: 69
Expires: 120
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY, OPTIONS, PRACK
Contact: <sip:[email protected]>
Supported: 100rel
Content-Length: 450
Content-Type: application/sdp

v=0
o=Nokia-SIPUA 2147483647 2147483647 IN IP4 10.28.185.239
s=-
c=IN IP4 10.28.185.239
t=0 0
m=audio 2000 RTP/AVP 8 18 96 97 98 13
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 AMR/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 telephone-event/8000
a=rtpmap:13 CN/8000
a=sendrecv
a=rtcp:2001 IN IP4 10.28.185.239
a=ptime:20
a=maxptime:200
a=fmtp:18 annexb=no
a=fmtp:96 mode-change-neighbor=1
a=fmtp:98 0-15

Source: 10.28.186.50:5060
Destination: 10.28.175.64:5080

SIPFlowMessage 2 - captured: 06/03/2009 17:37:51.323 CEST
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.28.186.50:5060;branch=z9hG4bK-076e792d779e21aeea4edc3df0892a05
Via: SIP/2.0/UDP 10.28.185.239:5060;branch=z9hG4bK37r32rgn6dhc6cs3no8iq8n
From: <sip:1075@Netphoneserver>;tag=jtqj2rhqc9hc7d3ano8k
To: <sip:1430@Netphoneserver;user=phone>;tag=3ac00400
Call-ID: FXINrGd5oIccbHUhL3BAaa0uo5ellK
CSeq: 955 INVITE
Contact: "Username"<sip:[email protected]:5080;user=phone>
Content-Length: 0


Source: 10.28.175.64:5080
Destination: 10.28.186.50:5060

SIPFlowMessage 3 - captured: 06/03/2009 17:37:53.263 CEST
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.28.186.50:5060;branch=z9hG4bK-076e792d779e21aeea4edc3df0892a05
Via: SIP/2.0/UDP 10.28.185.239:5060;branch=z9hG4bK37r32rgn6dhc6cs3no8iq8n
From: <sip:1075@Netphoneserver>;tag=jtqj2rhqc9hc7d3ano8k
To: <sip:1430@Netphoneserver;user=phone>;tag=3ac00400
Call-ID: FXINrGd5oIccbHUhL3BAaa0uo5ellK
CSeq: 955 INVITE
Contact: "Username"<sip:[email protected]:5080;user=phone>
Content-Length: 205
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, TRANSFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Type: application/sdp

v=0
o=- 3453032273 3453032273 IN IP4 10.28.175.64
s=e-phone session
c=IN IP4 10.28.175.64
t=3453032273 0
m=audio 6004 RTP/AVP 8 98
a=rtpmap:8 PCMA/8000
a=rtpmap:98 telephone-event/8000
a=ptime:20

Source: 10.28.175.64:5080
Destination: 10.28.186.50:5060

SIPFlowMessage 4 - captured: 06/03/2009 17:37:53.417 CEST
ACK sip:[email protected]:5080;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.28.185.239:5060;branch=z9hG4bKa0klqafclqfbdvfq65n8s53;rport
To: <sip:1430@Netphoneserver;user=phone>;tag=3ac00400
From: <sip:1075@Netphoneserver>;tag=jtqj2rhqc9hc7d3ano8k
Call-ID: FXINrGd5oIccbHUhL3BAaa0uo5ellK
CSeq: 955 ACK
Supported: sec-agree
Max-Forwards: 70
Content-Length: 0


Source: 10.28.185.239:5060
Destination: 10.28.175.64:5080

SIPFlowMessage 5 - captured: 06/03/2009 17:37:53.437 CEST
BYE sip:[email protected]:5080;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.28.185.239:5060;branch=z9hG4bKg3pj2riihthc6fgq05k1aeo;rport
To: <sip:1430@Netphoneserver;user=phone>;tag=3ac00400
From: <sip:1075@Netphoneserver>;tag=jtqj2rhqc9hc7d3ano8k
Call-ID: FXINrGd5oIccbHUhL3BAaa0uo5ellK
CSeq: 956 BYE
Max-Forwards: 70
Content-Length: 0


Source: 10.28.185.239:5060
Destination: 10.28.175.64:5080

SIPFlowMessage 6 - captured: 06/03/2009 17:37:53.778 CEST
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.28.185.239:5060;branch=z9hG4bKg3pj2riihthc6fgq05k1aeo
From: <sip:1075@Netphoneserver>;tag=jtqj2rhqc9hc7d3ano8k
To: <sip:1430@Netphoneserver;user=phone>;tag=3ac00400
Call-ID: FXINrGd5oIccbHUhL3BAaa0uo5ellK
CSeq: 956 BYE
Content-Length: 0


Source: 10.28.175.64:5080
Destination: 10.28.185.239:5060

SIPFlowMessage 7 - captured: 06/03/2009 17:37:55.276 CEST
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.28.186.50:5060;branch=z9hG4bK-076e792d779e21aeea4edc3df0892a05
Via: SIP/2.0/UDP 10.28.185.239:5060;branch=z9hG4bK37r32rgn6dhc6cs3no8iq8n
From: <sip:1075@Netphoneserver>;tag=jtqj2rhqc9hc7d3ano8k
To: <sip:1430@Netphoneserver;user=phone>;tag=3ac00400
Call-ID: FXINrGd5oIccbHUhL3BAaa0uo5ellK
CSeq: 955 INVITE
Contact: "Username"<sip:[email protected]:5080;user=phone>
Content-Length: 205
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, TRANSFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Type: application/sdp

v=0
o=- 3453032273 3453032273 IN IP4 10.28.175.64
s=e-phone session
c=IN IP4 10.28.175.64
t=3453032273 0
m=audio 6004 RTP/AVP 8 98
a=rtpmap:8 PCMA/8000
a=rtpmap:98 telephone-event/8000
a=ptime:20

Source: 10.28.175.64:5080
Destination: 10.28.186.50:5060

Hat jemand das entsprechende Know-How um mir anhand des Traces sagen zu können was das Problem sein könnte? Oder hat jemand ähnliche Probleme mit einem Voip-Gateway in Verbindung mit Nokias Standard-SIP-Client?

Danke schon mal und Gruss
 
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