asterisk users - Jun 2011 - Outgoing calls get dropped on high-latency connections.

We're a VoIP provider essentially competing with our local incumbent  
Telco, and a sizeable number of our customers use satellite internet.  
As a result, these customers never have ping times less than 500ms,  
and are often exceeding 2500ms.

I manually apply a "patch" to the Asterisk source code every time we  
upgrade Asterisk, described here:  
http://www.mail-archive.com/asterisk-users at lists.digium.com/msg178034.html  
This change allows our satellite customers to maintain their SIP  
connection for more than 5 minutes. But we're currently using Asterisk  
1.6.2.17, and this version seems to have one very strange bug on these  
high latency connections. On outgoing and *only* outgoing calls, the  
call drops after two or three minutes. Incoming calls do not have this  
problem, so I don't think it's the SIP connection getting killed due  
to a slow INVITE response.

Has anyone heard of this bug? Or should I submit a new bug report to  
the Asterisk project?

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Have you tried SIP session timer values in sip.conf

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ernie Dunbar
Sent: Tuesday, June 28, 2011 9:33 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Outgoing calls get dropped on high-latency
connections.

We're a VoIP provider essentially competing with our local incumbent Telco,
and a sizeable number of our customers use satellite internet.  
As a result, these customers never have ping times less than 500ms, and are
often exceeding 2500ms.

I manually apply a "patch" to the Asterisk source code every time we
upgrade
Asterisk, described here:  
http://www.mail-archive.com/asterisk-users at lists.digium.com/msg178034.html
This change allows our satellite customers to maintain their SIP connection
for more than 5 minutes. But we're currently using Asterisk 1.6.2.17, and
this version seems to have one very strange bug on these high latency
connections. On outgoing and *only* outgoing calls, the call drops after two
or three minutes. Incoming calls do not have this problem, so I don't think
it's the SIP connection getting killed due to a slow INVITE response.

Has anyone heard of this bug? Or should I submit a new bug report to the
Asterisk project?

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This message was sent using Lightspeed.ca's Advanced Webmail.



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