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.
------------------------------------------------------------------------
---------------
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@17q.com]
Sent: 04 October 2004 10:29
To: asterisk-users@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@17q.com
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