man, 18.02.2008 kl. 09.26 +0100, skrev CtRiX:
> Mikael Aleksander Bjerkeland wrote:
> > I tried nat=never. No difference. The logs from this call can be found
> > here: http://openpbx.pastebin.ca/908268
> >
> > I will try to do some regresssion testing to find out where it broke.
> >
> Useless logs.
>
> Please:
> sip debug,
> set verbose 9
> set debug 9
> rtp debug
>
> and use the full log, not the console.
Log: http://openpbx.pastebin.ca/908318
>
> Logs from the other party may be useful as well.
The other party (server1) shows the following (I could not easily debug
as there were lots of other calls at the same time):
<-- SIP read from 10.100.4.191:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.100.4.191:5060;branch=z9hG4bK5ad4245b;rport
From: "Mikael" <sip:[EMAIL PROTECTED]>;tag=as001b498d
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Date: Mon, 18 Feb 2008 09:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 214
v=0
o=root 3938 3938 IN IP4 10.100.4.191
s=session
c=IN IP4 10.100.4.191
t=0 0
m=audio 17472 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (13 headers 10 lines)---
Using INVITE request as basis request -
[EMAIL PROTECTED]
Sending to 10.100.4.191 : 5060 (NAT)
Found peer 'server2-server1'
Reliably Transmitting (no NAT) to 10.100.4.191:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.100.4.191:5060;branch=z9hG4bK5ad4245b;rport;received=10.100.4.191
From: "Mikael" <sip:[EMAIL PROTECTED]>;tag=as001b498d
To: <sip:[EMAIL PROTECTED]>;tag=as21825a13
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: cns
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Proxy-Authenticate: Digest realm="domain.com", nonce="3ee53ae4"
Content-Length: 0
---
Scheduling destruction of call
'[EMAIL PROTECTED]' in 15000 ms
server1*CLI>
<-- SIP read from 10.100.4.191:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.100.4.191:5060;branch=z9hG4bK5ad4245b;rport
From: "Mikael" <sip:[EMAIL PROTECTED]>;tag=as001b498d
To: <sip:[EMAIL PROTECTED]>;tag=as21825a13
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0
--- (10 headers 0 lines)---
server1*CLI>
<-- SIP read from 10.100.4.191:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.100.4.191:5060;branch=z9hG4bK19d39d7f;rport
From: "Mikael" <sip:[EMAIL PROTECTED]>;tag=as001b498d
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Proxy-Authorization: Digest username="server2-server1",
realm="domain.com", algorithm=MD5,
uri="sip:[EMAIL PROTECTED]", nonce="3ee53ae4",
response="daa66981b1c95cf1a11ac5a0624cb2bf", opaque=""
Date: Mon, 18 Feb 2008 09:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 214
v=0
o=root 3938 3939 IN IP4 10.100.4.191
s=session
c=IN IP4 10.100.4.191
t=0 0
m=audio 17472 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (14 headers 10 lines)---
Using INVITE request as basis request -
[EMAIL PROTECTED]
Sending to 10.100.4.191 : 5060 (NAT)
Found peer 'server2-server1'
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.100.4.191:17472
Peer video RTP is at port 10.100.4.191:65535
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0
(nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 11221122 in dialthrough (domain server1.domain.com)
list_route: hop: <sip:[EMAIL PROTECTED]>
Transmitting (no NAT) to 10.100.4.191:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.100.4.191:5060;branch=z9hG4bK19d39d7f;rport;received=10.100.4.191
From: "Mikael" <sip:[EMAIL PROTECTED]>;tag=as001b498d
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: cns
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
---
-- Executing Dial("SIP/server2-server1-e0ff", "Zap/g1/11221122") in
new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/11221122
We're at 10.100.4.190 port 17108
Video is at 10.100.4.190 port 17014
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 10.100.4.191:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
10.100.4.191:5060;branch=z9hG4bK19d39d7f;rport;received=10.100.4.191
From: "Mikael" <sip:[EMAIL PROTECTED]>;tag=as001b498d
To: <sip:[EMAIL PROTECTED]>;tag=as3e135e6b
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: cns
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 216
v=0
o=root 27673 27673 IN IP4 10.100.4.190
s=session
c=IN IP4 10.100.4.190
t=0 0
m=audio 17108 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Sent RTP packet to 10.100.4.191:17472 (type 8, seq 30812, ts 160, len
160)
Sent RTP packet to 10.100.4.191:17472 (type 8, seq 30813, ts 320, len
160)
Sent RTP packet to 10.100.4.191:17472 (type 8, seq 30814, ts 480, len
160)
Sent RTP packet to 10.100.4.191:17472 (type 8, seq 30815, ts 640, len
160)
Sent RTP packet to 10.100.4.191:17472 (type 8, seq 30816, ts 800, len
160)
Sent RTP packet to 10.100.4.191:17472 (type 8, seq 30817, ts 960, len
160)
Sent RTP packet to 10.100.4.191:17472 (type 8, seq 30818, ts 1120, len
160)
-- Zap/2-1 is proceeding passing it to SIP/server2-server1-e0ff
Sent RTP packet to 10.100.4.191:17472 (type 8, seq 30819, ts 1280, len
160)
Sent RTP packet to 10.100.4.191:17472 (type 8, seq 30820, ts 1440, len
160)
...
...
...
Sent RTP packet to 10.100.4.191:17472 (type 8, seq 32164, ts 216480, len
160)
Sent RTP packet to 10.100.4.191:17472 (type 8, seq 32165, ts 216640, len
160)
Feb 18 10:47:22 NOTICE[27696]: chan_sip.c:11500 do_monitor:
Disconnecting call 'SIP/server2-server1-e0ff' for lack of RTP activity
in 21 seconds
-- Hungup 'Zap/2-1'
-- Hungup 'Zap/2-1'
set_destination: Parsing <sip:[EMAIL PROTECTED]> for address/port to
send to
set_destination: set destination to 10.100.4.191, port 5060
Reliably Transmitting (no NAT) to 10.100.4.191:5060:
BYE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.100.4.190:5060;branch=z9hG4bK13726079;rport
From: <sip:[EMAIL PROTECTED]>;tag=as3e135e6b
To: "Mikael" <sip:[EMAIL PROTECTED]>;tag=as001b498d
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 BYE
User-Agent: cns
Max-Forwards: 70
Content-Length: 0
---
server1*CLI>
<-- SIP read from 10.100.4.191:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.100.4.190:5060;branch=z9hG4bK13726079;received=10.100.4.190;rport=5060
From: <sip:[EMAIL PROTECTED]>;tag=as3e135e6b
To: "Mikael" <sip:[EMAIL PROTECTED]>;tag=as001b498d
Call-ID: [EMAIL PROTECTED]
CSeq: 102 BYE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
X-CallWeaver-HangupCause: Normal Clearing
--- (12 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
As you can see from the logs on server2 (the ones at pastebin) it is not
"reliably transmitting (NAT)", but server1 is. Why is that? Server2 is
not sending any RTP traffic to server1 as you can see from the RTP
debug.
>
> Max
Apparently something else seems to be causing this. I reverted to 4015
and the problem is the same. Odd, since the configuration hasn't
changed. The only thing I changed was CallWeaver and SpanDSP. Could the
problem have been introduced in SpanDSP?
Are the logs I provided good enough? Any ideas why the behaviour
changed?
>
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