[asterisk-dev] streaming Asterisk audio online

[Steve Kann]

On Sep 20, 2006, at 3:58 AM, Tim Panton wrote:

>
> On 19 Sep 2006, at 15:26, Christian Croft wrote:
>
>> Can anyone point me towards some resources for streaming Asterisk  
>> phone call channels live on the web?  Has anyone looked into doing  
>> this, or is it even possible?
>
> We have a java applet that does it via an IAX call direct to asterisk,
> this is good where you care about low-latency but all the users
> will have their own asterisk channels.
>
> Otherwise you will need to look at embedding an audio client in the  
> page
> (eg quicktime/flash/real/WindowsMediaPlayer/winamp) and a server that
> goes from SIP/IAX to their streaming media -
> I seem to remember some integration into icecast was mentioned a  
> while back.
> You will get much higher latency with this sort of set up - lots of  
> buffers
> involved - but probably more scalable.

playSIP is a tool that can take a SIP call (gsm or uLaw), and re- 
broadcast it to an RTSP/RTP server, where RTSP/RTP clients like  
QuickTime can play it.    This kind of setup can get you in the  
3-5sec latency range, and very good scalability.