[asterisk-users] New thread - SIP over VPN

[Frank Bulk]

Depending on the latency, wrapping the UDP stream into a TCP-based tunnel
can be good -- if the VPN tunnel occasionally drops a packet, the tunnel
will re-transmit the UDP packet.  Of course, if the (one-way) latency is too
high, the re-transmitted payload will arrive outside the jitter buffer and
be dropped by the SIP CPE.

Frank

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Saturday, September 26, 2009 2:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] New thread - SIP over VPN


On Sat, 26 Sep 2009, Alan Lord (News) wrote:

>
> Hmmm, has anyone tried SIP over a VPN?
>
> We are thinking of testing this but haven't yet...
>
> Al
>

I have a client with Sonicwall VPNs.  Asterisk is at head office on 
internal LAN, six external locations all have Linksys 2102 ATAs and 
Polycom IP501 phones registering and placing calls through the tunnels. It 
seems to work fine, but there is plenty of bandwidth between the offices, 
and they use G729.  I think wrapping up the UDP stream into a TCP based 
tunnel might cause havoc if there is any packet loss or delay.

j

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