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
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--
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
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