--- Uriel Carrasquilla <[EMAIL PROTECTED]> wrote:
 John:
 are you aware of any documentation on how to configre SER to be a
 front-end
 to Asterisk?
 I suspect it is very inexpensive to put a SER server in a hosting
 facility

I think the cost is about the same as for putting a web server at a hosting facility. But I don't think you need high bandwidth. SER simply sets up the call. I don't think the audio data actually goes through SER. It goes directly between the two end points.

This is the big problem with using Asterisk for SIP. With Asterisk
the audio data between two SIP extensions has to actualy go into
then out of the Asterisk box. This does not scale well to
thousands of users like in a university campus or a comercial
SIP service.


[snip]

No, not always. If you leave "reinvite" permission turned on, Asterisk will supposedly send the audio between the two SIP endpoints. However, if NAT is in the equation, you're out of luck, since there needs to be an external media router that can translate between the two endpoints. If you choose to do clever things like use the "t" or "T" dial options, then you cannot release the media away from Asterisk since the system needs to listen to the RTP stream for cues.

Personally, I have had bad experimental luck with getting Asterisk to release media streams between two SIP endpoints. I can't say it's not possible, but I'll say that one of Asterisk's greatest features is it's media conversion routines (physical conversion, as well as codec and protocol) so I rarely have need to allow the media stream to move natively between endpoints, anyway.

However, in a large environment where the IP layer is more of a 'tree' than a ring, and the likely interconnection between peers within that local network is high, and where there are a large number of end stations, then SER or some other SIP proxy makes far more sense than Asterisk as a core call router.

JT
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