Hi there,
I have a running FreePBX Distro on a Vsphere Cluster. 2x Xeon 2620v3 16GB DDR3.
I have some problems with audio delay.
My SIP Provider uses G.711
I use G.722 for local calls.
When i call out with G.711 Codec everything is working find.
When i call out with G.722, Asterisk is Transcoding to G.711. But i have a delay from 0.5 to 1 Second. This is much too long.
My Audio Layer Latency with Transcoding from G722 -> G711 is arround 580ms
Any Tips, solutions? I dont know how to solve this
What are your trunk settings on the sip settings tab? Particularly the settings disallow and allow?
For example, I have in the PEER Details in the sip Settings tab…
disallow=all
allow=ulaw&g729
Also, check your Extension settings, Advanced page.
What is the Disallowed Codecs set to? (Mine = all)
What is the Allowed Codecs set to? (Mine = g722&ulaw&g729)
Might seem like a silly question. But how do you know your transcoding times? You said you used core show transcode. However, are you actually experiencing delay on those extensions when dialing out?
Sorry i meaned core show translation not transcoding 15ms
In X-Lite i can open a Device Test during a call. there i can see the codec which is used for the call and X-Lite also says i have an Audio Layer Latency 580ms with g722.
When i call with alaw, X-Lite says i have only 22ms Latency.