I know WCS can forward audio from a SIP call to a RTMP server. Does it also allow to forward audio data directly to an open remote TCP port? Or alternatively send it over websockets?
Thank you, but I don't see any attached files
I need to save (e.g. record) audio from a sip call in original format, G.711 in my case. I need to record chunks real-time, that's why I'm looking into rtmp streaming. My only issue is I need original audio format, not converted one.
I'm using SIP as RTMP feature to forward audio data from a SIP call to RTMP server. However it seems that audio is pre-converted as .flv. Is there a way to get raw audio (which is u-law encoded in my case) directly from WCS or probably on RTMP server side?
I’m trying to forward data from a SIP call to my RTMP server. Everything seems to work except I don’t see the stream on my server side. WCS is successfully connecting to a call, and also judging from the logs (please see attachment) it is able to publish RTMP stream. My codecs in...