[PROBLEM] SNOM 360 mit QSC: Request Timeout wenn RTP Verschlüsselung "an"

der_Gersthofer · 29 Mai 2005

Zwar kommt jetzt bei QSC kein Server Error (vgl. dazu anderer Thread) und ich bin auch am QSC Server angemeldet, jedoch: nach der Wahl der Rufnummer kommt jetzt ein "Request timeout".
Merkwürdig ist dabei insbes:
Warning: 392 213.148.130.46:5060 "Noisy feedback tells: pid=3304 req_src_ip=213.148.136.66 req_src_port=11018 in_uri=sip:[email protected]:5060;user=phone out_uri=sip:[email protected]:5060;user=phone via_cnt==1"

Code:

Sent to udp:213.148.136.2:5060 at 29/5/2005 13:15:00:190 (1079 bytes):
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.121:2054;branch=z9hG4bK-r8peck1k7ajg;rport
From: "Name - QSC" <sip:[email protected]>;tag=08n7yhqye7
To: <sip:[email protected];user=phone>
Call-ID: 3c2781c0cf85-9zp3vnfk4osw@snom360
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:[email protected]:2054;line=0y9zeldr>
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/3.60i
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 370

v=0
o=root 1929782907 1929782907 IN IP4 192.168.0.121
s=call
c=IN IP4 192.168.0.121
t=0 0
m=audio 51654 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

Received from udp:192.168.0.1:5060 at 29/5/2005 13:15:00:320 (355 bytes):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.121:2054;branch=z9hG4bK-r8peck1k7ajg;received=192.168.0.121;rport=2054
From: "Name - QSC" <sip:[email protected]>;tag=08n7yhqye7
Call-ID: 3c2781c0cf85-9zp3vnfk4osw@snom360
CSeq: 1 INVITE
Server: Intertex ix66-3-1-5---rel-3-15 
To: <sip:[email protected];user=phone>
Content-Length: 0

Received from udp:192.168.0.1:5060 at 29/5/2005 13:15:00:370 (748 bytes):
SIP/2.0 407 Proxy Authentication Required
From: "Name - QSC" <sip:[email protected]>;tag=08n7yhqye7
To: <sip:[email protected];user=phone>;tag=a91a18d8
CSeq: 1 INVITE
Call-ID: 3c2781c0cf85-9zp3vnfk4osw@snom360
Via: SIP/2.0/UDP 192.168.0.121:2054;rport=2054;branch=z9hG4bK-r8peck1k7ajg;received=192.168.0.121
Proxy-Authenticate: Digest realm="qsc.de",
nonce="4299a56022b0ca4cbe4cc79f6507ef5ee3178333",
qop="auth"
Server: Sip EXpress router (0.10.99-dev0 (i386/linux))
Warning: 392 213.148.130.46:5060 "Noisy feedback tells: pid=3304 req_src_ip=213.148.136.66 req_src_port=11018 in_uri=sip:[email protected]:5060;user=phone out_uri=sip:[email protected]:5060;user=phone via_cnt==1"
Content-Length: 0

Sent to udp:213.148.136.2:5060 at 29/5/2005 13:15:00:380 (406 bytes):
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.121:2054;branch=z9hG4bK-r8peck1k7ajg;rport
From: "Name - QSC" <sip:[email protected]>;tag=08n7yhqye7
To: <sip:[email protected];user=phone>;tag=a91a18d8
Call-ID: 3c2781c0cf85-9zp3vnfk4osw@snom360
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:[email protected]:2054;line=0y9zeldr>
Content-Length: 0

Sent to udp:213.148.136.2:5060 at 29/5/2005 13:15:00:390 (1336 bytes):
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.121:2054;branch=z9hG4bK-dy3t3r1fo2vb;rport
From: "Name - QSC" <sip:[email protected]>;tag=08n7yhqye7
To: <sip:[email protected];user=phone>
Call-ID: 3c2781c0cf85-9zp3vnfk4osw@snom360
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:[email protected]:2054;line=0y9zeldr>
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/3.60i
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Proxy-Authorization: Digest username="0821.....",realm="qsc.de",nonce="4299a56022b0ca4cbe4cc79f6507ef5ee3178333",uri="sip:[email protected];user=phone",qop=auth,nc=00000001,cnonce="485dfd75",response="53a599bbb462275298aef3628c7d20ce",algorithm=md5
Content-Type: application/sdp
Content-Length: 370

v=0
o=root 1929782907 1929782907 IN IP4 192.168.0.121
s=call
c=IN IP4 192.168.0.121
t=0 0
m=audio 51654 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

Received from udp:192.168.0.1:5060 at 29/5/2005 13:15:00:430 (355 bytes):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.121:2054;branch=z9hG4bK-dy3t3r1fo2vb;received=192.168.0.121;rport=2054
From: "Name - QSC" <sip:[email protected]>;tag=08n7yhqye7
Call-ID: 3c2781c0cf85-9zp3vnfk4osw@snom360
CSeq: 2 INVITE
Server: Intertex ix66-3-1-5---rel-3-15 
To: <sip:[email protected];user=phone>
Content-Length: 0

Received from udp:192.168.0.1:5060 at 29/5/2005 13:15:31:420 (388 bytes):
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 192.168.0.121:2054;branch=z9hG4bK-dy3t3r1fo2vb;received=192.168.0.121;rport=2054
From: "Name - QSC" <sip:[email protected]>;tag=08n7yhqye7
Call-ID: 3c2781c0cf85-9zp3vnfk4osw@snom360
CSeq: 2 INVITE
Server: Intertex ix66-3-1-5---rel-3-15 
To: <sip:[email protected];user=phone>;tag=1817512315149074648
Content-Length: 0

Sent to udp:213.148.136.2:5060 at 29/5/2005 13:15:31:430 (417 bytes):
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.121:2054;branch=z9hG4bK-dy3t3r1fo2vb;rport
From: "Name - QSC" <sip:[email protected]>;tag=08n7yhqye7
To: <sip:[email protected];user=phone>;tag=1817512315149074648
Call-ID: 3c2781c0cf85-9zp3vnfk4osw@snom360
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:[email protected]:2054;line=0y9zeldr>
Content-Length: 0

der_Gersthofer · 30 Mai 2005

Im Status findet sich
[email protected]: OK

Im LOG
[2]30/5/2005 22:42:53: Registered at registrar as [email protected]

Wenn ich aber die Nummer von außen anruft erhalte ich die Meldung: "Die von Ihnen gewählte Rufnummer ist zur Zeit nicht vergeben."

Mit dem QSC-Client kann ich dagegen telefonieren.

der_Gersthofer · 6 Jun 2005

Also, angerufen kann ich nun auf der Nr. werden.
Aber bei abgehenden Telefonaten kommt immer noch der Request Timeout.

JUN 06 21:06:25 tSip info recv 192.168.0.121: INVITE sip:[email protected];user=phone SIP/2.0
JUN 06 21:06:25 t5 info send sf 213.148.136.2: INVITE sip:[email protected];user=phone SIP/2.0
JUN 06 21:06:25 t2 info send sf 192.168.0.121: SIP/2.0 100 Trying
JUN 06 21:06:25 tSip info recv 213.148.136.2: SIP/2.0 100 Trying
JUN 06 21:06:25 tSip info recv 213.148.136.2: SIP/2.0 407 Proxy Authentication Required
JUN 06 21:06:25 t5 info send sf 213.148.136.2: ACK sip:[email protected];user=phone SIP/2.0
JUN 06 21:06:25 t2 info send sf 192.168.0.121: SIP/2.0 407 Proxy Authentication Required
JUN 06 21:06:25 tSip info recv 192.168.0.121: ACK sip:[email protected];user=phone SIP/2.0
JUN 06 21:06:25 tSip info recv 192.168.0.121: INVITE sip:[email protected];user=phone SIP/2.0
JUN 06 21:06:25 t5 info send sf 213.148.136.2: INVITE sip:[email protected];user=phone SIP/2.0
JUN 06 21:06:25 t2 info send sf 192.168.0.121: SIP/2.0 100 Trying
JUN 06 21:06:26 tSip info Invalid data received at SIP port
JUN 06 21:06:26 t5 info send sf 213.148.136.2: INVITE sip:[email protected];user=phone SIP/2.0
JUN 06 21:06:28 t5 info send sf 213.148.136.2: INVITE sip:[email protected];user=phone SIP/2.0
JUN 06 21:06:32 t5 info send sf 213.148.136.2: INVITE sip:[email protected];user=phone SIP/2.0
JUN 06 21:06:36 tSip info Invalid data received at SIP port
JUN 06 21:06:40 t5 info send sf 213.148.136.2: INVITE sip:[email protected];user=phone SIP/2.0
JUN 06 21:06:47 tSip info Invalid data received at SIP port
JUN 06 21:06:56 tSip error No answer from destination 213.148.136.2:5060
JUN 06 21:06:56 tSip info Remembering bad SIP server 213.148.136.2:5060.
JUN 06 21:06:56 tSip info No destination SIP server(s) left to fall back to.
JUN 06 21:06:56 t2 info send sf 192.168.0.121: SIP/2.0 408 Request Timeout
JUN 06 21:06:56 tSip info recv 192.168.0.121: ACK sip:[email protected];user=phone SIP/2.0

[hr:e286fe45e2]

Mit einem SIPURA geht es dagegen:

JUN 06 21:32:25 tSip info recv 192.168.0.125: INVITE sip:[email protected] SIP/2.0
JUN 06 21:32:25 t5 info send sf 213.148.136.2: INVITE sip:[email protected] SIP/2.0
JUN 06 21:32:25 t1 info send sf 192.168.0.125: SIP/2.0 100 Trying
JUN 06 21:32:25 tSip info recv 213.148.136.2: SIP/2.0 100 Trying
JUN 06 21:32:25 tSip info recv 213.148.136.2: SIP/2.0 407 Proxy Authentication Required
JUN 06 21:32:25 t5 info send sf 213.148.136.2: ACK sip:[email protected] SIP/2.0
JUN 06 21:32:25 t1 info send sf 192.168.0.125: SIP/2.0 407 Proxy Authentication Required
JUN 06 21:32:25 tSip info recv 192.168.0.125: ACK sip:[email protected] SIP/2.0
JUN 06 21:32:25 tSip info recv 192.168.0.125: INVITE sip:[email protected] SIP/2.0
JUN 06 21:32:25 t5 info send sf 213.148.136.2: INVITE sip:[email protected] SIP/2.0
JUN 06 21:32:25 t1 info send sf 192.168.0.125: SIP/2.0 100 Trying
JUN 06 21:32:26 tSip info recv 213.148.136.2: SIP/2.0 100 Trying
JUN 06 21:32:28 tSip info recv 213.148.136.2: SIP/2.0 180 Ringing
JUN 06 21:32:28 t1 info send sf 192.168.0.125: SIP/2.0 180 Ringing
JUN 06 21:32:31 tSip info Invalid data received at SIP port
JUN 06 21:32:37 tSip info recv 213.148.136.2: SIP/2.0 200 OK
JUN 06 21:32:37 tSip info SIP state(0x89ce94c) = ON_GOING
JUN 06 21:32:37 tSip info SIP state(0x89d3954) = ON_GOING
JUN 06 21:32:37 t1 info send sf 192.168.0.125: SIP/2.0 200 OK
JUN 06 21:32:37 tSip info recv 192.168.0.125: ACK sip:[email protected]:5060;user=phone SIP/2.0
JUN 06 21:32:37 t5 info send sf 213.148.136.2: ACK sip:[email protected]:5060;user=phone SIP/2.0
JUN 06 21:32:41 tSip info Invalid data received at SIP port
JUN 06 21:32:51 tSip info Invalid data received at SIP port
JUN 06 21:33:02 tSip info Invalid data received at SIP port
JUN 06 21:33:18 tSip info Invalid data received at SIP port
JUN 06 21:33:28 tSip info Invalid data received at SIP port
JUN 06 21:33:38 tSip info Invalid data received at SIP port
JUN 06 21:33:41 tSip info recv 192.168.0.125: BYE sip:[email protected]:5060;user=phone SIP/2.0
JUN 06 21:33:41 tSip info SIP state(0x89d3954) = DISCONNECTING
JUN 06 21:33:41 tSip info SIP state(0x89ce94c) = DISCONNECTING
JUN 06 21:33:41 t5 info send sf 213.148.136.2: BYE sip:[email protected]:5060;user=phone SIP/2.0
JUN 06 21:33:41 tSip info SIP state(0x89d11a0) = DISCONNECTING
JUN 06 21:33:41 tSip info recv 213.148.136.2: SIP/2.0 200 OK
JUN 06 21:33:41 t1 info send sf 192.168.0.125: SIP/2.0 200 OK

stefan_voip · 30 Dez 2005

Hallo,

dasselbe Problem hab' ich auch (snom 360 mit 5.0er Firmware), aber nur, wenn ich von meinem ipfonie-Account andere ipfonie-Festnetznummern anrufen möchte. Mit x_lite oder so kein Problem, nur mit dem snom 360 klemmt's.
Ebenso kann ich nicht von einer FritzBox 5050 mein Snom 360 anrufen (auch 408er Timeout-Fehler), wenn bei beiden eine ipfonie-Rufnummer erfolgreich registriert ist.

Hast Du das Problem inzwischen schon lösen können oder hat sonst noch jemand Probleme damit?

Grüße
Stefan

der_Gersthofer · 30 Dez 2005

Liegt an QSC

http://www.qdsl-support.de/showthread.php?t=10706

stefan_voip · 30 Dez 2005

Hallo,

den Thread hatte ich auch schon gelesen, hoffen wir mal, dass das Update seitens QSC bald kommt...

Nur komischerweise kann ich ja alle sonstigen Rufnummern anrufen ohne Probleme. Mit der snom 4.3er Firmware konnte ich sogar schon mal eine FritzBox ipfonie-intern anrufen - umgekehrt gings allerdings nicht.

Grüße
Stefan

der_Gersthofer · 4 Feb 2006

Nachdem das Problem nach einem Factory Reset wieder aufgetaucht ist:

Erst nachdem ich in der betreffenden Leitung RTP Verschlüsselung auf "aus" gesetzt hatte, war zumindest dieses Problem umschifft...

der_Gersthofer · 10 Feb 2006

Scheinbar sendet QSC irgendwelchen Datenmurks...

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[PROBLEM] SNOM 360 mit QSC: Request Timeout wenn RTP Verschlüsselung "an"

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