[asterisk-dev] streaming Asterisk audio online

[Tim Panton]

On 20 Sep 2006, at 16:53, Steve Kann wrote:

>
> 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.

Yep, whereas straight IAX gets you to the ~100ms mark.

In some cases this makes no difference, but for things like auctions,  
betting, voting
or other time sensitive things it makes _all_ the difference, it just  
depends
on your application.


Tim Panton

www.mexuar.com/cards.html