Alfred E. Heggestad
Alfred E. Heggestad
are you able to provide a wireshark trace ?
can you check if the function `check_telev()` is called while sending DTMF ? https://github.com/baresip/baresip/blob/main/src/audio.c#L557
Thanks for checking We are using the audio source to pace out DTMF packets, mainly for accurate timing. this is perhaps not a good solution ... You could try some...
aha, that is great, then you can use "thread" mode and it should be working. the alternative is to add a new timer that is started when a DTMF tone...
what I mean is, what type of product is your remove SIP endpoint ? can you verify that it receives the DTMF tones with good accuracy ?
if your endpoint is working well with baresip, please consider add it to the Interop table here: https://github.com/baresip/baresip/wiki/Interoperability for the DTMF part I dont think we can find a easy...
I will close this one now as there is a workaround, perhaps we will find a better solution later. Thanks for updating the interop table.
thanks for the report would you like to create a PR for this ?
Please have a look at this RFC: https://www.rfc-editor.org/rfc/rfc7160.html Another audio codec that has a difference between clock-rate, is G722. The SDP says g722/8000 but the actual clockrate for audio samples...
here is a proposal for how to implement this: 1. Create a list of all sampling rates used for all codecs. 2. For each sampling rate in that list, add...