I here have an OXO with version ONEDE032/039.001. The device works well with G711.
Anybody got G.722 working? Or the other way around, why isn't our OXO negotiate codec G722?
I made a dump with a test call incoming from my mobile. I attached 2 pictures from the dump, the invite, where you can see there are a lot of codecs offered. And the second the corresponding 200/ok reply. There is just codec 8 (G711a) listed.
So in the SIP-Gateway settings is of course G722 selected. Anywhere for me to look at? Maybe I don't understand that topic not in total but it makes no sense to me. Why isn't that codec negotiated?
Best regards
Alex
(I'm just an IT guy, who want's to understand.)
G.722 working?
G.722 working?
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Re: G.722 working?
G722 only works for IP phones (80X8 or 80X8S, and SIP-Phones). rtp_direct must be enabled in your SIP gateway media configuration and has to be configured in subscriber IP/SIP for voice coding/decoding.
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Demonslayer4000
- Member
- Posts: 19
- Joined: 13 Aug 2012 21:35
- Location: Australia
Re: G.722 working?
Hello Alex,
G722 is only available on some handsets and will only work with RTP direct and codec pass-through enabled on the SIP Trunks.
Then the codecs available on the handsets will be sent through to the Carrier to establish G722.
The issues you may have, Voicemail, announcements and MOH are all going to be either G711 or G729 as the OXO DSP's do not support G722. If the carrier does not like switching codecs once a call is established this might cause either calls or audio to drop out.
The second consideration is with RTP Direct, if the network is not designed properly and the handsets arent configured with the correct Gateway then the RTP will not be able to get routed directly to the SIP carrier. The other problem comes from any remote handsets as they will send RTP to the local router which will send the packets to the carrier from a different WAN IP, the carrier may reject the RTP Packets causing audio loss.
I hope this helps a little
Cheers
G722 is only available on some handsets and will only work with RTP direct and codec pass-through enabled on the SIP Trunks.
Then the codecs available on the handsets will be sent through to the Carrier to establish G722.
The issues you may have, Voicemail, announcements and MOH are all going to be either G711 or G729 as the OXO DSP's do not support G722. If the carrier does not like switching codecs once a call is established this might cause either calls or audio to drop out.
The second consideration is with RTP Direct, if the network is not designed properly and the handsets arent configured with the correct Gateway then the RTP will not be able to get routed directly to the SIP carrier. The other problem comes from any remote handsets as they will send RTP to the local router which will send the packets to the carrier from a different WAN IP, the carrier may reject the RTP Packets causing audio loss.
I hope this helps a little
Cheers
OXO ACSE | OXO Trainer
Omniswitch ACFE
OXE ACSE
Omniswitch ACFE
OXE ACSE
Re: G.722 working?
Thank you both for your answers. This was the solution, which unfortunately our service partner could not tell me.
Yes, there is still a small problem with the routing for devices in other networks, but I can certainly solve that. Thanks again, great job.
Best regards
Alex
Yes, there is still a small problem with the routing for devices in other networks, but I can certainly solve that. Thanks again, great job.
Best regards
Alex
