Hi there,
I originally posted under the topic of getting the codec for IVR menus changed from L16@8000 to L16@16000 in order to change the bitrate of the recordings that were played.
I posted an update there saying that I had found out that the codec for the IVR is selected based on the codec of the call, and indeed when I change my codec negotiation from "scrooge" to "generous" I saw that the codec appropriate for the call was being selected.
It seemed at the time as though the quality had improved, than I was getting fewer "chops" and "cutouts", however today doing more testing of IVR menus, the voicemail system, and recordings, I'm finding that on every call, at least once, there is a "choppy bit" or a "cutout"; that is, a momentary silence or jump in the audio. The quality is clear, but these little skips/chops are always there.
It's not a complete show stopper, but certainly if I'm adding customers with their IVR menus I want the quality of the outbound audio to be at least as good as the quality of the calls in general, which run without any skips/chops.
When I do a pcap and listen to the audio, the skips/chops are present, so this appears to be something that is happening on the FreeSWITCH server rather than something that is happening on the network between the SIP client or upstream gateway and the FS server.
Does anyone have any clues about how to improve the quality of recordings/wavs being played?
Thanks!
Iain
I originally posted under the topic of getting the codec for IVR menus changed from L16@8000 to L16@16000 in order to change the bitrate of the recordings that were played.
I posted an update there saying that I had found out that the codec for the IVR is selected based on the codec of the call, and indeed when I change my codec negotiation from "scrooge" to "generous" I saw that the codec appropriate for the call was being selected.
It seemed at the time as though the quality had improved, than I was getting fewer "chops" and "cutouts", however today doing more testing of IVR menus, the voicemail system, and recordings, I'm finding that on every call, at least once, there is a "choppy bit" or a "cutout"; that is, a momentary silence or jump in the audio. The quality is clear, but these little skips/chops are always there.
It's not a complete show stopper, but certainly if I'm adding customers with their IVR menus I want the quality of the outbound audio to be at least as good as the quality of the calls in general, which run without any skips/chops.
When I do a pcap and listen to the audio, the skips/chops are present, so this appears to be something that is happening on the FreeSWITCH server rather than something that is happening on the network between the SIP client or upstream gateway and the FS server.
Does anyone have any clues about how to improve the quality of recordings/wavs being played?
Thanks!
Iain