opus codec

[Roy Reynolds]

Hello all, 

I'm trying to register a Fanvil extension with opus codec and it fails.  I made sure it's enabled on the PBx and on the phone.  Is there something else that I should do? 

Thanks for the responses. 

Regards,

[OnRelay Support]

We are happy take a look if you could share some registrar / proxy error logs & failure symptoms.

[Roy Reynolds]

Thanks. I will gather that information and send it as soon as I'm able to. 

Best regards, 

--
You received this message because you are subscribed to the Google Groups "sipxcom-users" group.
To unsubscribe from this group and stop receiving emails from it, send an email to sipxcom-user...@googlegroups.com.
To view this discussion visit https://groups.google.com/d/msgid/sipxcom-users/63931042-16d2-4067-a2ea-ecb4369a2806n%40googlegroups.com.

[Matt Keys]

Registration has nothing to do with codecs in use.

Typically registration issues are due to DNS failing to resolve the DNS SRV records for the SIP domain. Make sure your Fanvil is looking to sipx (or a DNS server that has all the SRV records for your SIP domain that sipx has) for the primary DNS server.

Codec negotiation happens during the 200 OK with SDP and ACK to that. If opus isn't available on one side or the other, they will negotiate another codec that both has available. Sipx isn't involved in this process, unless you're attempting to use a B2BUA function on sipx. Those are basically any freeswitch service -- things like IVR / AA, conference or voicemail. It's also important to understand that you're not going to be able to use opus on the PSTN. You'd only be able to use that on internal calls between extension to extension.

Regards,

Matt

[Roy Reynolds]

Thanks for your detailed answer Matt. 

I have struggled to understand why it fails the registration when I select opus as the codec. If I change only the codec to a G.711 one it registers. The DNS server is the sipx itself. 

Best Regards. 

--

You received this message because you are subscribed to the Google Groups "sipxcom-users" group.
To unsubscribe from this group and stop receiving emails from it, send an email to sipxcom-user...@googlegroups.com.

To view this discussion visit https://groups.google.com/d/msgid/sipxcom-users/226f63d2-10d0-40c6-9718-99c787b51fc3n%40googlegroups.com.

[Matt Keys]

Without seeing the log or pcap of the REGISTER attempt, I can only guess... but a good guess may be you're using UDP transport on the Fanvil rather than TCP. UDP has a 1500 byte size limitation. Adding codecs adds in extra lines to the message, which may have pushed you over that 1500 byte limitation. Use TCP instead to get around that limitation. 

Regards,

Matt

[Roy Reynolds]

Thanks Matt. I thought the very same thing and tried TCP and it still didn't register. I am going to try again and save the pcap this time. I was using tcpdump to watch the traffic real time..

Regards,

[Support]

It may depend on how you are defining the Fanvil phone in sipX phone admin. 

Some of the legacy phone definitions have a valid codecs list.

If you try to define it as a CounterPath Bria “opus” is included as a valid codec in the definitions there.

To view this discussion visit https://groups.google.com/d/msgid/sipxcom-users/544cccc6-abde-47ee-ab87-3b94e14f1aa8%40Spark.

[Roy Reynolds]

Thanks for your response. 

 

I did define it as a Bria phone. I made sure opus was included in the available list of codecs.  I know the phone negotiated on all available codecs during the registration process because it picks one of the G711 ones and registers. If I however remove all the other codecs and only leave opus, it doesn't register. 

 

I'm trying to have high definition audio for internal calls only. 

Regards,

[Support]

The answer here will likely be found in the sipX logs for the registrar and / or the proxy.  Look for any exception, error, or similar.

[Roy Reynolds]

Happy to report that after struggling with this for awhile, I finally got the fanvil phone to register with OPUS as the only codec.  That's the good news.  The bad news is that on the version of SIPXECS none of the media profiles have OPUS as a codec to add so it can't call to AutoAttendant or a conference Bridge.

Thanks for all the help and suggestions though.  They helped me crack this one.

[Matt Keys]

You'd need to add it into the freeswitch enabled codecs under media services for AA and conference use. 

You received this message because you are subscribed to a topic in the Google Groups "sipxcom-users" group.
To unsubscribe from this topic, visit https://groups.google.com/d/topic/sipxcom-users/txzdIk97fWM/unsubscribe.
To unsubscribe from this group and all its topics, send an email to sipxcom-user...@googlegroups.com.
To view this discussion visit https://groups.google.com/d/msgid/sipxcom-users/67f04f59-57c2-4aae-98f4-ac14a7f5d3b9n%40googlegroups.com.

[Donkiss Boss]

Licensed Pharmaceuticals And Psychedelics Online Store
Buy the best Pharmaceuticals And Psychedelics products in our website with safe and guaranteed shipping and delivery Worldwide
Check out our website below if interested in our products.
https://shorturl.at/BvDVf
https://shorturl.at/ITBJW
https://shorturl.at/wCGS4
https://shorturl.at/RnP0b
https://shorturl.at/GTr8Q