aram
2005-Mar-19 20:30 UTC
[Asterisk-Users] Problem Making a SIP call over a long latency network - Call rejected: 407 Proxy Authentication Required
Hello,
We are getting error: Call rejected: 407 Proxy Authentication Required - if
a user is trying to call using * over a long latency network (around 600
ms). There is no problem when the same user is trying to make a call with
low latency network (around 300 ms). I have included the debug and log
messages for Asterisk. This call is done with SJphone, the same problem
exists with ATA; however X-Pro is having no problem. Similarly if the user
is not authenticated - the call goes through fine.
Is * timing out waiting for some response - if so how can we
increase the timeout? We are also thinking of possibility that some port is
closed over the long latency network that causing this problem - is that
possible -if yes, which port would that be?
Or is there another issue that we are not aware of?
Thanks,
Aram
Debug (some parts of it)
======================
Sip read:
INVITE sip:2002@xxx.xxx.xxx.xxx SIP/2.0
Content-Length: 360
Contact: <sip:2000@82.198.1.15:5060>
Call-ID: 959E39B1-B5BA-4F76-952B-192A9E4829EF@82.198.1.15
Content-Type: application/sdp
From: "2000"<sip:2000@xxx.xxx.xxx.xxx>;tag=608598751280
CSeq: 1 INVITE
Max-Forwards: 70
To: <sip:2002@xxx.xxx.xxx.xxx>
Via:
SIP/2.0/UDPyyy.yyy.yyy.yyy;rport;branch=z9hG4bK52c6010f0131c9b14237ee5f00007
9bb00000045
User-Agent: SJLabs-SJphone/1.40.258
====
12 headers, 16 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 503 Unavailable
Via:
SIP/2.0/UDPyyy.yyy.yyy.yyy;branch=z9hG4bK52c6010f000000244237ee60000052da000
00047
From: "2000"<sip:2000@xxx.xxx.xxx.xxx>;tag=608598751280
To: <sip:2002@xxx.xxx.xxx.xxx>;tag=as0338b9e1
Call-ID: 959E39B1-B5BA-4F76-952B-192A9E4829EF@82.198.1.15
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2002@xxx.xxx.xxx.xxx>
Content-Length: 0
========
toyyy.yyy.yyy.yyy:5060
In the log
Mar 18 07:32:37 DEBUG[2325]: Setting NAT on RTP to 4 Mar 18 07:32:38
NOTICE[2325]: Unable to create/find channel Mar 18 07:32:38 DEBUG[2325]:
Stopping retransmission on '5CE9576F-92DB-4C4D-928E-DD704559AB38@82.198
Mar 18 07:32:38 DEBUG[2325]: Setting NAT on RTP to 4 Mar 18 07:32:38
DEBUG[2325]: Ignoring too old packet packet 1 (expecting >= 2) Mar 18
07:32:38 NOTICE[2325]: Unable to create/find channel
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aram
2005-Mar-22 01:43 UTC
[Asterisk-Users] Problem Making a SIP call over a long latency network - Call rejected: 407 Proxy Authentication Required
Hello,
We are getting error: Call rejected: 407 Proxy Authentication Required - if
a user is trying to call using * over a long latency network (around 600
ms). There is no problem when the same user is trying to make a call with
low latency network (around 300 ms). I have included the debug and log
messages for Asterisk. This call is done with SJphone, the same problem
exists with ATA; however X-Pro is having no problem. Similarly if the user
is not authenticated - the call goes through fine.
Is * timing out waiting for some response - if so how can we
increase the timeout? We are also thinking of possibility that some port is
closed over the long latency network that causing this problem - is that
possible -if yes, which port would that be?
Or is there another issue that we are not aware of?
Thanks,
Aram
Debug (some parts of it)
======================
Sip read:
INVITE sip:2002@xxx.xxx.xxx.xxx SIP/2.0
Content-Length: 360
Contact: <sip:2000@82.198.1.15:5060>
Call-ID: 959E39B1-B5BA-4F76-952B-192A9E4829EF@82.198.1.15
Content-Type: application/sdp
From: "2000"<sip:2000@xxx.xxx.xxx.xxx>;tag=608598751280
CSeq: 1 INVITE
Max-Forwards: 70
To: <sip:2002@xxx.xxx.xxx.xxx>
Via:
SIP/2.0/UDPyyy.yyy.yyy.yyy;rport;branch=z9hG4bK52c6010f0131c9b14237ee5f00007
9bb00000045
User-Agent: SJLabs-SJphone/1.40.258
====
12 headers, 16 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 503 Unavailable
Via:
SIP/2.0/UDPyyy.yyy.yyy.yyy;branch=z9hG4bK52c6010f000000244237ee60000052da000
00047
From: "2000"<sip:2000@xxx.xxx.xxx.xxx>;tag=608598751280
To: <sip:2002@xxx.xxx.xxx.xxx>;tag=as0338b9e1
Call-ID: 959E39B1-B5BA-4F76-952B-192A9E4829EF@82.198.1.15
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2002@xxx.xxx.xxx.xxx>
Content-Length: 0
========
toyyy.yyy.yyy.yyy:5060
In the log
Mar 18 07:32:37 DEBUG[2325]: Setting NAT on RTP to 4 Mar 18 07:32:38
NOTICE[2325]: Unable to create/find channel Mar 18 07:32:38 DEBUG[2325]:
Stopping retransmission on '5CE9576F-92DB-4C4D-928E-DD704559AB38@82.198
Mar 18 07:32:38 DEBUG[2325]: Setting NAT on RTP to 4 Mar 18 07:32:38
DEBUG[2325]: Ignoring too old packet packet 1 (expecting >= 2) Mar 18
07:32:38 NOTICE[2325]: Unable to create/find channel
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Craig
2005-Mar-22 02:07 UTC
[Asterisk-Users] Problem Making a SIP call over a long latency network - Call rejected: 407 Proxy Authentication Required
Hi Aram,
Think you will find it is something to do with ports etc....
We log soft phones, spa-841, grandstream and sipura ata through a
satellite link that regularly drifts out to a latency of 1000ms ++ and
do not see any problems with *.
However some of the links we have used are into port blocking and
stateful packet blocking. Suggest this would be most likely your cause.
If you can, run it through a vpn over the same link and see if the
problem persists, this will make life very simple.
For those that think a 100ms is long latency, I have seen our link drift
out to 1200ms latency with 100ms of jitter and still work, does start to
get a bit crappy though!
Craig.
Message: 28
Date: Tue, 22 Mar 2005 00:43:35 -0800
From: "aram" <aram@hi-teck.com>
Subject: [Asterisk-Users] Problem Making a SIP call over a long
latency network - Call rejected: 407 Proxy Authentication
Required
To: <asterisk-users@lists.digium.com>,
<asterisk-dev@lists.digium.com>
Message-ID: <auto-000003091938@hi-teck.com>
Content-Type: text/plain; charset="us-ascii"
Hello,
We are getting error: Call rejected: 407 Proxy Authentication Required -
if
a user is trying to call using * over a long latency network (around 600
ms). There is no problem when the same user is trying to make a call
with
low latency network (around 300 ms). I have included the debug and log
messages for Asterisk. This call is done with SJphone, the same problem
exists with ATA; however X-Pro is having no problem. Similarly if the
user
is not authenticated - the call goes through fine.
Is * timing out waiting for some response - if so how can we
increase the timeout? We are also thinking of possibility that some
port is
closed over the long latency network that causing this problem - is that
possible -if yes, which port would that be?
Or is there another issue that we are not aware of?
Thanks,
Aram