Hi,

I always wonder what is the benefit to use VoIP proxy server like OpenSER ?
- what is the benefit of it ? ( I am not sure to see this )
Why not to use a simple Asterisk server can do the job ?

I can see on thir website : ( http://opensips.org )
OpenSIPS (Open SIP Server) is a mature Open Source implementation of a
SIP server. OpenSIPS is more than a SIP proxy/router as it includes
application-level functionalities. OpenSIPS, as a SIP server, is the
core component of any SIP-based VoIP solution. With a very flexible and
customizable routing engine, OpenSIPS 'unifies voice, video, IM and
presence services in a highly efficient way, thanks to its scalable
(modular) design.

- Should it be the solution of all the problems, that we can have with
Router, NAT etc.... ?

Thanks for your help

On 27/08/2008 10:58, Steve wrote:

I always wonder what is the benefit to use VoIP proxy server like OpenSER ?
Why not to use a simple Asterisk server can do the job ?

Asterisk is not a SIP proxy, if for example you receive an VoIP inbound
call, and want to divert it to a VoIP extension, all the RTP traffic
will pass through asterisk for the duration of the call.

I believe openSER can setup the call during the SIP negotiation phase so
that the calling party directly contacts the diverted extension once the
call is answered and RTP data starts flowing.

http://www.voip-info.org/wiki-Asterisk+SIP+not-proxy

However if you used openSER you wouldn't have the DTMF/IVR processing
that asterisk would give you, as you wouldn't be "hearing" the RTP
traffic at all ..

Andy Burns wrote:
On 27/08/2008 10:58, Steve wrote:

I always wonder what is the benefit to use VoIP proxy server like
OpenSER ?
Why not to use a simple Asterisk server can do the job ?

Asterisk is not a SIP proxy, if for example you receive an VoIP inbound
call, and want to divert it to a VoIP extension, all the RTP traffic
will pass through asterisk for the duration of the call.

I believe openSER can setup the call during the SIP negotiation phase so
that the calling party directly contacts the diverted extension once the
call is answered and RTP data starts flowing.

http://www.voip-info.org/wiki-Asterisk+SIP+not-proxy

However if you used openSER you wouldn't have the DTMF/IVR processing
that asterisk would give you, as you wouldn't be "hearing" the RTP
traffic at all ..



Thanks for your reply.

I just had a look on Voip-Info. They also mentioned some other SIP proxies.

Do you mean, that with SIP proxy, we can more easily manage RTP packages ??
What is the benefit ? ( I can manage RTP packages , for an optimisation
I prioritized the ports difnined into my RTP.conf .... )

Steve wrote:
Andy Burns wrote:
On 27/08/2008 10:58, Steve wrote:

I always wonder what is the benefit to use VoIP proxy server like
OpenSER ?
Why not to use a simple Asterisk server can do the job ?
Asterisk is not a SIP proxy, if for example you receive an VoIP inbound
call, and want to divert it to a VoIP extension, all the RTP traffic
will pass through asterisk for the duration of the call.

I believe openSER can setup the call during the SIP negotiation phase so
that the calling party directly contacts the diverted extension once the
call is answered and RTP data starts flowing.

http://www.voip-info.org/wiki-Asterisk+SIP+not-proxy

However if you used openSER you wouldn't have the DTMF/IVR processing
that asterisk would give you, as you wouldn't be "hearing" the RTP
traffic at all ..



Thanks for your reply.

I just had a look on Voip-Info. They also mentioned some other SIP proxies.

Do you mean, that with SIP proxy, we can more easily manage RTP packages ??
What is the benefit ? ( I can manage RTP packages , for an optimisation
I prioritized the ports difnined into my RTP.conf .... )


Just a precision...

Yes, you are right. THe communicaiton will go through Asterisk IF the
Asterisk server are behind a NAT. If not, The RTP packets, can go
directly through the other SIP phone without going to Asterisk first

- Can I have OpenSER WIHTOUT Asterisk ???? ( If I can register to
OpenSER, no need to register to Asterisk ?!!?

Steve wrote:
- Can I have OpenSER WIHTOUT Asterisk ???? ( If I can register to
OpenSER, no need to register to Asterisk ?!!?


Yes.

Tim

On Wed, 27 Aug 2008 17:21:34 +0100, Andy Burns wrote:

Asterisk is not a SIP proxy, if for example you receive an VoIP inbound
call, and want to divert it to a VoIP extension, all the RTP traffic
will pass through asterisk for the duration of the call.

http://www.voip-info.org/wiki-Asterisk+SIP+not-proxy

Oddly enough, that link you've posted doesn't mention that this behaviour
is configurable:

http://www.voip-info.org/wiki-Asterisk+sip+canreinvite

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