[OpenSIPS-Users] OpenSIPs as Registration server in front of Asterisk

[Todd Routhier]

Problem: Calls from PSTN provider > Asterisk > OpenSIPs > SIP Endpoint have
intermittent audio issues. See below for details.

I am a long time Asterisk user but extremely new to OpenSIPs.

We are in the process of a migration from an older Asterisk server to a
newer version along with some other changes.

First order of business is for us to offload all registrations from our
current 1.8.x Asterisk server to OpenSIPs 2.4.6.

We have a setup that seems to be mostly working but intermittent audio
issues are what we are trying to eliminate.

When I say intermittent, audio seems to work for a particular end point in
certain situations or it doesn't. For example, we have some end
points which have no audio at all such as my personal soft-phone. I can't
get audio on any of three different soft-phones on my laptop, no audio in
either direction. But, I have a Grandstream phone on the same LAN which
works perfectly every time, on every call.

I have other end points which are Grandstream phones with perfectly working
audio in both directions, all the time, consistently.

I have other Grandstream end points which work for the same type of call
every time, with audio in both directions but the same phone has no audio
on slightly different types of calls (hard to explain what I mean by "types
of calls").

Ideally, we would not care about this working with Asterisk 1.8.x since we
are moving away from it but it's important for it to work as part of our
transition/migration.

I had horrible audio issues at first were it was hardly working at all or
one way audio consistently. I fixed this by setting nat=yes in the sip.conf
for the context pointing to the OpenSIPs server. I couldn't understand why
this fixed it since the OpenSIPs server and the Asterisk server both have
static IP's and are NOT behind any NAT of any sort. Only the end points
registered to OpenSIPs are behind end points.

Still I noticed that Asterisk was trying to send calls to the LAN IP of the
end points, so I tested nat=yes and it fixed most of the audio issues with
only the issues outlined above remaining.

My next steps are to see if I have good audio if I push calls to the newer
Asterisk server then to the end points registered to the OpenSIPs server.
Even if that works, it does not solve my current need to make this work
with Asterisk 1.8.x at least until the migration is complete.

Thanks in advance for any assistance with this.

Regards,

Todd
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