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

[Todd Routhier]

Yes the end point phones are behind NAT but reach is behind a different
NAT. Typically one or two phones at each location. NATs are different at
each location so I'm sure this is why some work and some don't.

None of them use STUN but this is because I never had to use STUN for any
of these same runs points when registered directly to Asterisk.

On Wed, Oct 16, 2019, 2:07 AM Răzvan Crainea <razvan at opensips.org> wrote:

> Hi, Todd!
>
> Can you provide a pcap of one of the calls that are not working?
> Also, are these clients behind NAT? Do they use STUN?
>
> Best regards,
> Răzvan
>
> On 10/15/19 9:01 PM, Todd Routhier wrote:
> > 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
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
> --
> Răzvan Crainea
> OpenSIPS Core Developer
>    http://www.opensips-solutions.com
>
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