Hi Team, As checked there is no support of. multiple Codec support as i checked it only supports the ulaw format it should support all other codec format that are supported by audioSocket (websocket) i checked in asterisk 22.6.0 version
AudioSocket is not WebSocket-based. Also, it doesn’t support any actual codecs: the audio must be in PCM 16-bit little-endian format at 8000Hz sample rate.
so when we write using the ari connection for th e way for the external media we use the transport layer as the RTP and websocket so the websocket receives all the correct format for the codec support but the RTP session of the layer do not support the Codec support my main goal is to send the audio to a external interface with the all the available code support currently it only support the aulaw support
I’m not 100% sure what this thread really means, but the RTP support in external media is indeed limited to specific static payloads in RTP - as there is no negotiation of dynamic payloads. The websocket support allows all codecs. If you’re asking for the RTP support to have support for dynamic payloads so it can support more codecs, I’m not aware of anyone working on such a thing. You could submit a feature request[1] though no guarantee anyone will work on it.
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