[Asterisk-Dev] Multiple Asterisks on one machine
chris at netservers.co.uk
Tue Feb 17 05:10:49 MST 2004
> 1) To solve a different problem I was able quite easily to run 2 x
> Asterisk on one box, using UML (User mode linux).
Due to the extra latency added by using UML, I don't think this would be
an option, but nice idea.
> 2) You mention you are binding Asterisk to specific interfaces, however I
> think this only controls the receive side - have you considered the
> transmit side? My understanding is that the send and receive sides of a
> session are quite distinct. In my case I was trying to set up tunnels
> (PPTP,L2TP) and it proved impossible to control the used interfaces for
> transmitting in a multi-interface system, which in turn prevented
> tunnelling from working. (without getting into fixing the code ;-))
I think it depends on the software. I haven't looked in great detail
(yet) into whether the same (bound) socket is used for sending and
receiving by Asterisk, but I'm assuming for now that it is. In the tcpdump
trace you can see two distinct source IP addresses for Asterisk packets,
so it appears to be working.
> 3) Why is there a problem with CAPI and jitter buffer? hopefully on CAPI
> --> SIP direction the phone buffer should be effective? and surely the
> CAPI channel must provide some buffer otherwise it is hardly usefull?
The problem is in the SIP -> CAPI direction, because Asterisk, unlike SIP
phones, does not appear to have a jitter buffer (either in its own SIP
channel receiver, or in CAPI).
I too would have thought it essential, given that all SIP phones that I
know of have jitter buffers, even Grandstream. But I can't find it, and
deliberately introducing jitter really kills voice quality. Adding a
jitter buffer by linking two Asterisk boxes via IAX2 restores most of the
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