[Asterisk-Users] SIP Proxy and use with Asterisk

[Alex Barnes]

I did reply to these questions when you posted then under "HELP:
Asterisk - SIP to H.323 translation".

But not to worry probably got lost in the crowd that or my ramblings
aren't very helpful.

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---------------
Hi,

First Question:
This is rather a difficult question to answer.
Many people say that it allows * to scale better for SIP. For example
assuming your SIP proxy is stateful the proxy will handle all
retransmissions / redirections / register lookups / call logging / ????,
all of which would be hidden from the *.
Multiple registrations:
* doesn't support multiple registrations. A registrar proxy will be able
to do this, allowing such things as forking (sequential / parrellel /
combinations of both).

Grey Areas (Functionality that crosses over between * and a proxy)
-------------------------------------------------------------------
CPL:
Any proxy worth its salt will have a CPL engine built in allowing some
pretty powerful scripting of incoming / outgoing calls on a individual
user basis.
Application Servers:
A proxy may also be an application server. Which could be used to make
any number of technologies available. For example Ubiquity make a HA SIP
App Server that can do pretty much anything including RMI client/server
apps, SOAP, link with Web Application Servers (Websphere / Weblogic),
Java Eventlets that let you write your own interfaces using our SIP
SDK's for call handling.

HOWEVER!!!!!!!!!!!
----------------------
I don't think that Asterisk is quite ready to support all live
deployment scenarios that include a 3rd party SIP proxy. One problem I
ran into was Asterisk does not handle looped back calls. 
For example a call comes in over PSTN to Asterisk, Asterisk forwards to
your SIP registrar proxy, Registrar does a lookup on the SIP address and
finds that the user is register'd to an analogue phone. 
If the SIP registrar redirected using a 3xx response the * will play
along happily, but if the proxy wishes to stay in the loop (maybe you
have a billing application running on it) it would add a Record-Route
header to the SIP request , to say it wishes to receive all subsequent
messages for this call, and then proxy back to the *. The * will ignore
this INVITE totally. If the user had been registered to a proper SIP end
point then the loop back wouldn't have happened and this works a treat.

Second question:
Yes but looks like support isn't great / the community hasn't really
investigated this much (http://www.voip-info.org/wiki-Asterisk+video)
I plan to evaluate video over SIP next week / this weekend so if you get
anywhere with this please let me know. I will of course do likewise.

Third Question:

I don't know!


I hope this helps you.

Alex


-----Original Message-----
From: steve [mailto:steve at 17q.com] 
Sent: 04 October 2004 10:29
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] SIP Proxy and use with Asterisk


Hi Everyone:
 
I have a THREE questions.  What is a sip proxy and what is the benefit
of having one with Asterisk?   I am well aware that we have a sip
channel in Asterisk and that we have SIP registration.   I am not sure
why you would need a SIP server and OR a registration server.
  
 
Second question, with Asterisk are you able to do video on VOIP video
phones?
 
 
Last question, does iptel.org SIP Express Router work with Asterisk?
 
 
Thanks
 
 
Steve
steve at 17q.com


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