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

[Răzvan Crainea]

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