[Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)

[John Todd]

>--- Uriel Carrasquilla <uriel at adelphia.net> 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