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<TITLE>Re: [OpenSIPS-Users] Is opensips a front end to asterisk?</TITLE>
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<FONT FACE="Tahoma, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'>I’ll second that. I’m fairly new to Opensips, but not to SIP and definitely not to Asterisk. As I started to realize the limitations of Asterisk I looked for something a bit more powerful and flexible. I started reading some of the module documentation to get an idea what Opensips (Openser at the time) was capable of. It referenced a lot of things I didn’t understand. So, I ended up at RFC3261. It was invaluable to me to understand exactly what a proxy, uas, uac, and a few other key terms meant. And what their specific functions were in a SIP environment. I’m far from an expert but I’ve learned a lot. It has really helped me understand what a proxy’s role is, and also what a proxy’s role isn’t. Opensips is a proxy, after all.<BR>
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- Jeff<BR>
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On 7/9/09 1:53 PM, "Brett Nemeroff" <<a href="brett@nemeroff.com">brett@nemeroff.com</a>> wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT FACE="Tahoma, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'>I know this may sound like a pretty lame answer, but you'll get a lot of benefit from reading the definition of a SIP PROXY from RFC3261. <BR>
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You can't do much with OpenSIPS (properly) if you dont' know the underlying RFC. This is very different from other SIP software packages, like Asterisk where you pretty much can't break RFC compliance on purpose (hah, it may just already be broken)..<BR>
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-Brett<BR>
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2009/7/9 Raúl Alexis Betancor Santana <<a href="rabs@dimension-virtual.com">rabs@dimension-virtual.com</a>><BR>
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---------- Forwarded message ----------<BR>
From: "Raúl Alexis Betancor Santana" <<a href="rabs@dimension-virtual.com">rabs@dimension-virtual.com</a>><BR>
To: <a href="lists@grounded.net">lists@grounded.net</a><BR>
Date: Tue, 7 Jul 2009 20:47:01 +0100<BR>
Subject: Re: [OpenSIPS-Users] Is opensips a front end to asterisk?<BR>
On Tuesday 07 July 2009 20:15:01 <a href="lists@grounded.net">lists@grounded.net</a> wrote:<BR>
> On Tue, 7 Jul 2009 14:02:11 -0400, Alex Balashov wrote:<BR>
> > Specific and well-parameterised questions really are the key.<BR>
><BR>
> I'll certainly do that, once I start understanding the product but for now,<BR>
> I'm just trying to get a handle on basics, not deep in depth<BR>
> understandings, just enough to formulate a plan.<BR>
<BR>
You should go into deep knowleadge, it's a MUST to work with a sip proxy. You<BR>
could begin reading the "Getting starting gide" and so.<BR>
<BR>
> One was asking about the viability of using opensips on ESXi. Because of<BR>
> how easy it is to use snapshots, backup and so on, this would be the best<BR>
> working environment. So my question was, does opensips have any hardware<BR>
> timing requirement issues such as asterisk does. If timing is not critical,<BR>
> as a voip server is, then opensips must run nicely in a virtual manner.<BR>
<BR>
Yes, you could use it into a VM, no timming issues like Asterisk. Talking<BR>
about backup and so ... you only need to do a backup of the .cfg file and the<BR>
database backend you use, so you will not get any real advantage of running<BR>
inside a VM from backup point of view.<BR>
<BR>
> I don't have any numbers to work with, which is why I say scalable. I'm<BR>
> looking for something which can help me to scale a voip based application<BR>
> to many users. So let's say hundreds of users so that we have a number. I<BR>
> know many of you are running many thousands so this should be a good<BR>
> starting point.<BR>
<BR>
It depends on lot of variables, like available mem, CPU power, network (the<BR>
most important part), but also how complex is your .cfg about request<BR>
proccessing, how do you handle database request, etc.<BR>
So there is not a magic formula, but there are sip-proxies around the world<BR>
working with Million of users.<BR>
<BR>
> This is how I would have approached this, until I started looking for a sip<BR>
> gateway/load balancer.<BR>
<BR>
That's a setup, not direcly related with the software you use.<BR>
<BR>
> This should be pretty straight forward to those who have pro setups and<BR>
> want as much reliability as possible. I want to have two separate locations<BR>
> so that I can fail over, simple as that really.<BR>
<BR>
There are not "simple" scenarios in SIP world and faiolver is very-complex<BR>
one.<BR>
<BR>
> >-From what I can tell, opensips could act as a pbx on it's own but it can<BR>
> > act as a proxy/load >balancer/gateway to asterisk systems as well.<BR>
><BR>
> This is what I asked about in this thread a couple of times now. It's not<BR>
> fully clear to me, even after reading. It sometimes sounds like opensips<BR>
> can be a voip server though it does not provide other media services such<BR>
> as voice mail and so on. I get that it is a gateway but I'm trying to get a<BR>
> better understanding of FROM that point on.<BR>
<BR>
That's because Opensips it's a proxy, not a PBX, not a B2BUA, etc., it doesn't<BR>
manage media, so you need some "complements" to have a "full-featured VoIP<BR>
system"<BR>
<BR>
--<BR>
Raúl Alexis Betancor Santana<BR>
Dimensión Virtual<BR>
<BR>
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