[Asterisk-Users] HELP: Asterisk - SIP to H.323 translation

[Alex Barnes]

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.



I hope this helps you.

Alex



-----Original Message-----
From: steve [mailto:steve at 17q.com] 
Sent: 01 October 2004 11:14
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] HELP: Asterisk - SIP to H.323 translation


Hi:

I have a question.  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.   

Second question, with Asterisk are you able to do video on VOIP video
phones?

Thanks


Steve
steve at 17q.com



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Asterisk .
Sent: Thursday, September 30, 2004 4:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] HELP: Asterisk - SIP to H.323 translation

Hello,

--- UTRINI at embratel.com.br wrote:
<snip>

> Can I use Asterisk as a SIP Proxy and as a SIP to H.323 translator? I 
> want to implement  PC-to-Phone calls in the following topology (for
> signalling):
> SIP Softphone --> Asterisk --> Gatekeeper H.323 ---> Gateway H.323 
> ---> PSTN The RTP audio packets would go direct through Softphone to 
> gateway.

You can use Asterisk as a SIP-H323 translator. It is not a SIP proxy,
but a PBX having a SIP channel. It also is a SIP UAS/Registrar. I dont
think when it is used as a translator, RTP packets will go directly from
softphone to gateway, since there are 2 different protocols involved.
Asterisk will force the RTP packets to go through it.

> 
> Helaine
> 

Regards, Girish


		
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