RE: [Users] Issues with calls using openser.

3 Jan 2007

I changed the codec to ULAW which is defiantly supported.  I'm thinking the
reINVITE may be the problem but I'm pretty new at openser configuration and
don't see a clear way to detect a reinvite and not auth it.  I did capture a
ngrep of a failed call.  I'll also test xlite or a sipura tonight to see if
it something specific to the grandstream.

ngrep -q -t -W byline port 5060
interface: eth0 (192.168.16.0/255.255.255.0)
filter: (ip) and ( port 5060 )

U 2007/01/03 16:18:41.764242 192.168.16.91:5060 -> 192.168.16.192:5060
INVITE sip:7005874200@siprt1.siptest.net:5060;user=phone SIP/2.0.
t:   <sip:7005874200@siprt1.siptest.net:5060;user=phone>.
f:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
Remote-Party-Id:
<sip:7006311229@192.168.16.91:5060;user=phone>;screen=yes;id-type=subscriber
;party=calling;privacy=off.
Proxy-Require: privacy.
i: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979699 INVITE.
v: SIP/2.0/UDP 192.168.16.91:5060.
Max-Forwards: 70.
m: <sip:7006311229@192.168.16.91:5060;user=phone>.
k: replaces.
c: application/sdp.
Accept: application/sdp.
Accept-Encoding: .
Accept-Language: en.
User-Agent: MSTSYLVAIPGW.
l: 249.
.
v=0.
o=MSTNT 536702936 536702936 IN IP4 192.168.16.91.
s=Session SDP.
c=IN IP4 192.168.16.91.
t=0 0.
m=audio 40878 RTP/AVP 18 0 101 .
a=silenceSupp:off.
a=ecan:b on g168.
a=rtpmap:101 telephone-event/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.

U 2007/01/03 16:18:41.764646 192.168.16.192:5060 -> 192.168.16.91:5060
SIP/2.0 100 Giving a try.
t:   <sip:7005874200@siprt1.siptest.net:5060;user=phone>.
f:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
i: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979699 INVITE.
v: SIP/2.0/UDP 192.168.16.91:5060.
Server: OpenSer (1.2.0-dev12-notls (i386/linux)).
Content-Length: 0.
Warning: 392 192.168.16.192:5060 "Noisy feedback tells:  pid=14845
req_src_ip=192.168.16.91 req_src_port=5060
in_uri=sip:7005874200@siprt1.siptest.net:5060;user=phone
out_uri=sip:7005874200@192.168.17.83;user=phone via_cnt==1".
.

U 2007/01/03 16:18:41.764673 192.168.16.192:5060 -> 192.168.17.83:5060
INVITE sip:7005874200@192.168.17.83;user=phone SIP/2.0.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
t:   <sip:7005874200@siprt1.siptest.net:5060;user=phone>.
f:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
Remote-Party-Id:
<sip:7006311229@192.168.16.91:5060;user=phone>;screen=yes;id-type=subscriber
;party=calling;privacy=off.
Proxy-Require: privacy.
i: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979699 INVITE.
Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0.
v: SIP/2.0/UDP 192.168.16.91:5060.
Max-Forwards: 69.
m: <sip:7006311229@192.168.16.91:5060;user=phone>.
k: replaces.
c: application/sdp.
Accept: application/sdp.
Accept-Encoding: .
Accept-Language: en.
User-Agent: MSTSYLVAIPGW.
l: 249.
.
v=0.
o=MSTNT 536702936 536702936 IN IP4 192.168.16.91.
s=Session SDP.
c=IN IP4 192.168.16.91.
t=0 0.
m=audio 40878 RTP/AVP 18 0 101 .
a=silenceSupp:off.
a=ecan:b on g168.
a=rtpmap:101 telephone-event/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.

U 2007/01/03 16:18:41.774319 192.168.17.83:5060 -> 192.168.16.192:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0.
Via: SIP/2.0/UDP 192.168.16.91:5060.
From:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
To: <sip:7005874200@siprt1.siptest.net:5060;user=phone>.
Call-ID: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979699 INVITE.
User-Agent: Grandstream HT496 1.0.3.64 FXS0.
Content-Length: 0.
.

U 2007/01/03 16:18:41.776363 192.168.17.83:5060 -> 192.168.16.192:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0.
Via: SIP/2.0/UDP 192.168.16.91:5060.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
From:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
To:
<sip:7005874200@siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
Call-ID: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979699 INVITE.
User-Agent: Grandstream HT496 1.0.3.64 FXS0.
Content-Length: 0.
.

U 2007/01/03 16:18:41.776442 192.168.16.192:5060 -> 192.168.16.91:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.16.91:5060.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
From:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
To:
<sip:7005874200@siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
Call-ID: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979699 INVITE.
User-Agent: Grandstream HT496 1.0.3.64 FXS0.
Content-Length: 0.
.

U 2007/01/03 16:18:44.426461 192.168.17.83:5060 -> 192.168.16.192:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0.
Via: SIP/2.0/UDP 192.168.16.91:5060.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
From:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
To:
<sip:7005874200@siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
Call-ID: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979699 INVITE.
User-Agent: Grandstream HT496 1.0.3.64 FXS0.
Contact: <sip:7005874200@192.168.17.83;user=phone>.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Type: application/sdp.
Supported: replaces.
Content-Length: 220.
.
v=0.
o=7005874200 8000 8000 IN IP4 192.168.17.83.
s=SIP Call.
c=IN IP4 192.168.17.83.
t=0 0.
m=audio 10000 RTP/AVP 18 101.
a=sendrecv.
a=rtpmap:18 G729/8000.
a=ptime:20.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.

U 2007/01/03 16:18:44.426613 192.168.16.192:5060 -> 192.168.16.91:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.16.91:5060.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
From:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
To:
<sip:7005874200@siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
Call-ID: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979699 INVITE.
User-Agent: Grandstream HT496 1.0.3.64 FXS0.
Contact: <sip:7005874200@192.168.17.83;user=phone>.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Type: application/sdp.
Supported: replaces.
Content-Length: 220.
.
v=0.
o=7005874200 8000 8000 IN IP4 192.168.17.83.
s=SIP Call.
c=IN IP4 192.168.17.83.
t=0 0.
m=audio 10000 RTP/AVP 18 101.
a=sendrecv.
a=rtpmap:18 G729/8000.
a=ptime:20.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.

U 2007/01/03 16:18:44.451021 192.168.16.91:5060 -> 192.168.16.192:5060
ACK sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0.
t:
<sip:7005874200@siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
f:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
i: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979699 ACK.
v: SIP/2.0/UDP 192.168.16.91:5060.
Max-Forwards: 70.
Route: <sip:7005874200@192.168.17.83;user=phone>.
User-Agent: MSTSYLVAIPGW.
l: 0.
.

U 2007/01/03 16:18:44.451221 192.168.16.192:5060 -> 192.168.17.83:5060
ACK sip:7005874200@192.168.17.83;user=phone SIP/2.0.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
t:
<sip:7005874200@siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
f:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
i: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979699 ACK.
Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.2.
v: SIP/2.0/UDP 192.168.16.91:5060.
Max-Forwards: 69.
User-Agent: MSTSYLVAIPGW.
l: 0.
.

U 2007/01/03 16:18:44.451328 192.168.16.91:5060 -> 192.168.16.192:5060
INVITE sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0.
t:
<sip:7005874200@siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
f:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
Remote-Party-Id:
<sip:7006311229@192.168.16.91:5060;user=phone>;screen=yes;id-type=subscriber
;party=calling;privacy=off.
Proxy-Require: privacy.
i: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979700 INVITE.
v: SIP/2.0/UDP 192.168.16.91:5060.
Max-Forwards: 70.
Route: <sip:7005874200@192.168.17.83;user=phone>.
m: <sip:7006311229@192.168.16.91:5060;user=phone>.
c: application/sdp.
Accept: application/sdp.
Accept-Encoding: .
Accept-Language: en.
User-Agent: MSTSYLVAIPGW.
l: 236.
.
v=0.
o=MSTNT 536702936 536702937 IN IP4 192.168.16.91.
s=Session SDP.
c=IN IP4 192.168.16.91.
t=0 0.
m=audio 40878 RTP/AVP 18 101.
a=silenceSupp:off.
a=ecan:b on g168.
a=ptime:20.
a=rtpmap:101 telephone-event/8000.
a=rtpmap:18 G729/8000.

U 2007/01/03 16:18:44.453398 192.168.16.192:5060 -> 192.168.16.91:5060
SIP/2.0 407 Proxy Authentication Required.
t:
<sip:7005874200@siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
f:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
i: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979700 INVITE.
v: SIP/2.0/UDP 192.168.16.91:5060.
Proxy-Authenticate: Digest realm="192.168.16.91",
nonce="459c1ee0731e3291a5704b9666ffded6acb20bb5".
Server: OpenSer (1.2.0-dev12-notls (i386/linux)).
Content-Length: 0.
Warning: 392 192.168.16.192:5060 "Noisy feedback tells:  pid=14844
req_src_ip=192.168.16.91 req_src_port=5060
in_uri=sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598
out_uri=sip:7005874200@192.168.17.83;user=phone via_cnt==1".
.

U 2007/01/03 16:18:44.471821 192.168.16.91:5060 -> 192.168.16.192:5060
ACK sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0.
t:
<sip:7005874200@siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
f:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
i: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979700 ACK.
v: SIP/2.0/UDP 192.168.16.91:5060.
Max-Forwards: 70.
User-Agent: MSTSYLVAIPGW.
l: 0.
.

U 2007/01/03 16:18:44.472045 192.168.16.91:5060 -> 192.168.16.192:5060
BYE sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0.
t:
<sip:7005874200@siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
f:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
i: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979701 BYE.
v: SIP/2.0/UDP 192.168.16.91:5060.
Max-Forwards: 70.
Route: <sip:7005874200@192.168.17.83;user=phone>.
Accept: application/sdp.
Accept-Encoding: .
Accept-Language: en.
User-Agent: MSTSYLVAIPGW.
l: 0.
.

U 2007/01/03 16:18:44.474251 192.168.16.192:5060 -> 192.168.17.83:5060
BYE sip:7005874200@192.168.17.83;user=phone SIP/2.0.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
t:
<sip:7005874200@siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
f:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
i: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979701 BYE.
Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bKd29b.7f5f9c52.0.
v: SIP/2.0/UDP 192.168.16.91:5060.
Max-Forwards: 69.
Accept: application/sdp.
Accept-Encoding: .
Accept-Language: en.
User-Agent: MSTSYLVAIPGW.
l: 0.
.

U 2007/01/03 16:18:44.506433 192.168.17.83:5060 -> 192.168.16.192:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bKd29b.7f5f9c52.0.
Via: SIP/2.0/UDP 192.168.16.91:5060.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
From:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
To:
<sip:7005874200@siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
Call-ID: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979701 BYE.
User-Agent: Grandstream HT496 1.0.3.64 FXS0.
Contact: <sip:7005874200@192.168.17.83;user=phone>.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Supported: replaces.
Content-Length: 0.
.

U 2007/01/03 16:18:44.506549 192.168.16.192:5060 -> 192.168.16.91:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.16.91:5060.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
From:
<sip:7006311229@192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
To:
<sip:7005874200@siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
Call-ID: 1068bc1b-1bb-1ffd6fd8(a)192.168.16.91.
CSeq: 12979701 BYE.
User-Agent: Grandstream HT496 1.0.3.64 FXS0.
Contact: <sip:7005874200@192.168.17.83;user=phone>.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Supported: replaces.
Content-Length: 0.
.

U 2007/01/03 16:18:47.652882 192.168.17.150:5060 -> 192.168.16.192:5060
..................

-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] 
Sent: Wednesday, January 03, 2007 7:51 AM
To: Shane Burrell
Cc: users(a)openser.org
Subject: Re: [Users] Issues with calls using openser.

The log shows that you challenge reINVITEs. MAybe this breaks the 
grandstream. Please try without challenging the reINVITE. If this helps, 
then it is a probably a grandstream bug.

But of course there is the question why the grandstream sends a reINVITE 
at all? This is often a codec problem.

Please also try to use other client (e.g. xlite) and post complete ngrep 
dumps: "ngrep -q -t -W byline port 5060"

regards
klaus

Shane Burrell wrote:
...
  The SIP UA was a grandstream ATA running the latest
stable firmware. Prior
...
  to upgrading to 1.1 and moving to mediaproxy it worked
well with the
 exception of good nat support which is why I would really like mediaproxy to
...
  work.  Is there anything I should look for in the sip
dialog to determine if
...
  the client, sip proxy, or the gateway is the culprit
on disconnecting the
 call?

 Thanks,

 Shane

 -----Original Message-----
 From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] 
 Sent: Wednesday, January 03, 2007 4:52 AM
 To: Shane Burrell
 Cc: users(a)openser.org
 Subject: Re: [Users] Issues with calls using openser.

 Maybe a bug in the caller's SIP client?

 regards
 klaus

 Shane Burrell wrote:
  I recently installed the latest version of
openser and this time used
 mediaproxy rather than rtpproxy. Everything seems to work but if a sip
 device is called the phone rings and is instantally disconnected and the  far
  end is left off-hook.  This worked before but I
did modify my script to  work
  with mediaproxy.  Below is the wireshark decode
of the sip messagining.  Any
> help or suggestions on where to look would be great.  Calls from the sip
> device works flawlessly. I am using a MaxTNT as the gateway.
>
>  
>
>  
>
> |Time     | 152.93.36.91      | siprt1.me.net| 152.93.37.83      |
>
> |22.031   |         INVITE SDP ( telephone-event)          |
> |SIP From: sip:8385021101@192.168.16.91:5060 To:sip: 8385024200@
> siprt1.me.net:5060
>
> |         |(5060)   ------------------>  (5060)   |                   |
>
> |22.031   |         100 Giving a try              | |SIP
...
 > Status
>
> |         |(5060)   <------------------  (5060)   |                   |
>
> |22.031   |                   |         INVITE SDP ( telephone-event)
> |SIP Request
>
> |         |                   |(5060)   ------------------>  (5060)   |
>
> |22.040   |                   |         100 Trying| |SIP
...
 > Status
>
> |         |                   |(5060)   <------------------  (5060)   |
>
> |22.042   |                   |         180 Ringing |SIP
...
 > Status
>
> |         |                   |(5060)   <------------------  (5060)   |
>
> |22.042   |         180 Ringing                   | |SIP
...
 > Status
>
> |         |(5060)   <------------------  (5060)   |                   |
>
> |25.244   |                   |         200 OK SDP ( telephone-event)
> |SIP Status
>
> |         |                   |(5060)   <------------------  (5060)   |
>
> |25.245   |         200 OK SDP ( telephone-event)          |
> |SIP Status
>
> |         |(5060)   <------------------  (5060)   |                   |
>
> |25.269   |         ACK       |                   | |SIP
...
 > Request
>
> |         |(5060)   ------------------>  (5060)   |                   |
>
> |25.269   |                   |         ACK       | |SIP
...
 > Request
>
> |         |                   |(5060)   ------------------>  (5060)   |
>
> |25.269   |         INVITE SDP ( telephone-event)          |
> |SIP From: sip: 8385021101@152.93.36.91:5060 To:sip: 8385024200@
> siprt1.me.net:5060
>
> |         |(5060)   ------------------>  (5060)   |                   |
>
> |25.270   |         407 Proxy Authentication Required          |
> |SIP Status
>
> |         |(5060)   <------------------  (5060)   |                   |
>
> |25.291   |         ACK       |                   | |SIP
...
 > Request
>
> |         |(5060)   ------------------>  (5060)   |                   |
>
> |25.291   |         BYE       |                   | |SIP
...
 > Request
>
> |         |(5060)   ------------------>  (5060)   |                   |
>
> |25.293   |                   |         BYE       | |SIP
...
 > Request
>
> |         |                   |(5060)   ------------------>  (5060)   |
>
> |25.326   |                   |         200 OK    | |SIP
...
 > Status
>
> |         |                   |(5060)   <------------------  (5060)   |
>
> |25.327   |         200 OK    |                   | |SIP
...
   Status

 |         |(5060)   <------------------  (5060)   |                   |

 Shane

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Klaus Darilion
nic.at

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RE: [Users] Issues with calls using openser.