Re: [Live-devel] playSIP and stdout file not readable

2013-12-31 Thread Plischke, Markus
Finlayson Gesendet: Dienstag, 31. Dezember 2013 10:19 An: LIVE555 Streaming Media - development & use Betreff: Re: [Live-devel] playSIP and stdout file not readable First, I assume that your file is non-empty - i.e., you actually received data :-) Assuming that your file is non-empty, then it *should

Re: [Live-devel] playSIP and stdout file not readable

2013-12-31 Thread Ross Finlayson
Oops, I misread your last email. No, of course VLC can't play the stream, because VLC doesn't include a SIP client. So forget that... But the fact remains that the data that you're receiving *should* contain PCM a-law audio (1 channel, 8 bits-per-sample, 8000 Hz), because that's what your ser

Re: [Live-devel] playSIP and stdout file not readable

2013-12-31 Thread Ross Finlayson
First, I assume that your file is non-empty - i.e., you actually received data :-) Assuming that your file is non-empty, then it *should* be containing PCM a-law audio, because that's what the server (Asterisk) reported. The fact that VLC (which uses the LIVE555 RTSP/RTP client library) was ab

Re: [Live-devel] playSIP and stdout file not readable

2013-12-31 Thread Plischke, Markus
46 An: LIVE555 Streaming Media - development & use Betreff: Re: [Live-devel] playSIP and stdout file not readable You appear to be receiving the audio correctly, so I suspect your problem is simply that you are not decoding it properly. The data in your file ("test.alaw") should be

Re: [Live-devel] playSIP and stdout file not readable

2013-12-31 Thread Ross Finlayson
You appear to be receiving the audio correctly, so I suspect your problem is simply that you are not decoding it properly. The data in your file ("test.alaw") should be PCM a-law audio, 1 channel, with a sampling frequency of 8000 Hz. You probably need to tell "ffmpeg" explicitly what kind of

[Live-devel] playSIP and stdout file not readable

2013-12-31 Thread Plischke, Markus
Hi, ich have a little problem with playSIP. My goal ist o call a asterisk server in the same network and be member in a confbridge an record everything. So far, everything is working except reading the recording: Command: /usr/local/bin/playSIP -a -A 8 -u user secret sip:3@172.16.16.53:5060 > /