I’m using SIP phones all set to uLaw on a closed LAN. All calls are extension to extension for this application and there are no cards or external PSTN, etc.
When a call is placed direct from phone to phone, or from phone to asterisk to phone the call setups and there’s very little latency or delay in the audio coming to/from each end of the call.
However, when one extension calls a queue, it’s placed in the queue and then the queue rings one of the logged in extensions (again this is all extension to extension). The extension picks up the call.
At this point, the RTP sets up and the queue member (recieving end) can hear the caller, but the caller cannot hear the queue member right away. it takes maybe 3-5 seconds, then the both sides work well.
I have tried this with various phones, soft phones, etc. They’re all using the same codec. I’ve seen other posts similar to this situation, but many are going through firewalls, etc. I don’t have any of that here (all LAN, in fact same switch for all devices).
I’ve tried AsteriskNOW 1.0.2 (latest) and 1.4.6 and 1.4.16. I have not tried 1.4.23 yet.