SIP calls audio sniffing

[mbedial]

Hi all,
we are researching the options to transcribe a voip call in real time.
Currently, we use flashphoner as SIP voip client to call asterisk, and asterisk makes the call to any destination.
As far as I know, according to Flahphoner docs, it supports to forward the stream of a sip voice call via RTMP. However, this possibility for transcribing in real time the call is not a good option. Does Flasphoner support any othe way to stream to a third party a voice call? Does it support an url that we could access to the call?

Thanks in advance.
Mario

 

[Max]

Hello

There are two APIs

1) Web SDK, Android SDK, iOS SDK, call() hangup() answer()
If the SIP call has been created using this API, then such a call can't be re-streamed to RTMP endoint.
WebRTC SIP Gateway features

2) REST API /call/startup {toStream: "stream1"} or /call/startup {rtmpUrl: "rtmp://localhost:1935/live", rtmpStream:"stream1"}
If you make such a call using this API, then SIP RTP incoming traffic only can be re-streamed to RTMP or as a regular stream.
Outgoing SIP RTP traffic won't be re-streamed.
Streaming video and SIP integration features

Therefore you can't simply connect to existing two-party SIP call to see or listen both parties in real-time.
As an option you can try to add this line in flashphoner.properties

record = /usr/local/FlashphonerWebCallServer/records

The WCS server should record WAV files as a result of recording. Therefore you can use recorded files as a source for transcribing.