Mid-call Codec renegotiation

[Alex Lake]

Just had a problem with a SIP trunk:

- A call that started off using PCMU
- After 15 mins a SIP session refresh was sent offering PCMU & G729
- Carrier switch thought that a change to G729 was a good idea
- 3CX didn't honour the change

I'm sorry not to have all the SIP messages in detail, but just wondering if anyone else has come across this, or if they've seen successful midcall audio codec renegotiation at all?

BTW - It can be resolved by only having one code enabled on the SIP trunk.

 

[leejor]

Well, the standards do allow a re-INVITE to change the initial INVITE "settings". See section 14 ...https://www.ietf.org/rfc/rfc3261.txt
It may be that your provider has some test enabled to determine that the call quality would be better with a lower rate codec. You could discuss this with them, perhaps it is an option that can be turned off, or, do as you suggest, remove the UN-desired CODECs from the list. After all, if you don't want to use them, why make them available in the first place?

 

[Alex Lake]

Well, the standards do allow a re-INVITE to change the initial INVITE "settings". See section 14 ...https://www.ietf.org/rfc/rfc3261.txt
It may be that your provider has some test enabled to determine that the call quality would be better with a lower rate codec. You could discuss this with them, perhaps it is an option that can be turned off, or, do as you suggest, remove the UN-desired CODECs from the list. After all, if you don't want to use them, why make them available in the first place?

Sure - I can stop it happening, but I was wondering if there's a bug in the latest version of 3CX that means that mid-call codec renegotiation doesn't work - as although what's happening is perhaps unusual, it shouldn't stop working.
Perhaps there's a Jira or similar for 3CX?