We implemented a call queue and we have a queue announcement recording that plays for the agent when the call is being passed to them as well as the reported hold time. The issue was not present in our testing but when we went live with the queue it seems on some calls when the hold time message is played it introduces a distortion on the audio only for the agent end (the caller audio is clear). This seems to clear up after about a minute or so. If we disable the reporting of the hold time there is no static.
I suspect there is some kind of audio codec mismatch that is being introduced on the call. Does anyone have any suggestions?
Which version of Asterisk and what phones are the affected agents using? This sounds like an effect you get when there is a discontinuity in the RTP timestamps, without a change in the source identity (SSRC), on some phones, at least with older versions of Asterisk.
At least in older versions, Asterisk doesn’t update SSRC to match the current source, but misuses the marker bit in the RTP, when the source changes. The marker bit indicates a good place to reset a jitter buffer, but doesn’t mandate that. Asterisk assumes the downstream system will re-initialise the jitter buffer.
The phones are Mitel M 5360 IP phones. The odd thing is the queue recording announcement we manually created which plays before always plays clear then the hold time recording may or may not play clear and if not when the call connects the audio will be a bit distorted and hard to understand but then generally corrects itself in 1-2 minutes but not always.