openser integration

zenny

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#1
I have read somewhere (it is here http://forums.digium.com/viewtopic.php?t=19903) that asterisk cannot handle more than 200 simultaneous calls and openser can upto 3000-4000 simultaneously.

Is OpenSER + RTProxy or Mediaproxy integration with elastix possible so that OpenSER provides the telephony and the asterisk provides the services?

Or that has already been accomplished in Elastix v1.0?<br><br>Post edited by: zenny, at: 2008/02/08 23:30
 

cowboy47

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#2
From what I gathered from that thread, it was more an ad for someone looking for a job rather than a statement of fact. I know that approximately 2 years ago, in Chicago, one of the developers made a presentation at the communication industry seminar and stated that if you had a server that was dual core, with 2-4GB of memory and sufficient hard drive space, that the system would be capable of handling up to 10,000 simulateneous calls.

From looking at the specs of openser, I see little differences between asterisk & openser.

C
 

zenny

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#3
Glad to know and look forward to ELX 1.0. Thanks

Could you give a pointer to the presentation you are referring to? Worth reading!
 

cowboy47

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#4
I did not keep the presentation. It was made back when Trixbox was the only thing going and they had just changed from Asterisk@Home to Trixbox (so this was a while back).

What I did do, was go the openser sight and take a look at the features that they tout. To be quite honest, I do not see anything that indicates that they have more capabilities than Asterisk. They have many of the same features but I do not know the interface as I have not installed it.

But also, do the math. You would be hard pressed to find a reasonably priced cisco router that will support 10+ E1's (10 E1's would give you 300 channels if you were just using 711 codecs). By using different codecs, you have the ability to support more calls because of the bandwidth but if it were me, dealing in a call center environment, I would want to distribute that type of load over several servers.

C
 

zenny

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#5
I read this in one of the mailinglists:
http://www.colug.net/pipermail/colug432 ... 04960.html

[COLUG] Asterisk vs. OpenSER
Duane duane at e164.org
Sat Oct 27 18:11:05 EDT 2007

* Previous message: [COLUG] Asterisk vs. OpenSER
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Jeff Frontz wrote:

> It took me about 15 minutes to get asterisk up and running for my
> (admittedly very simple) need; it's taken hours (and HOURS) to get
> OpenSER set up (and I'm still not confident I have it right, since I
> can't reliably reproduce the desired behavior). Part of this, I'm sure,
> is because OpenSER is more scalable (I configured a user in Asterisk
> using a flat file, while I have to use encapsulated database operations
> via the openser daemon manipulated via commands exchanged with a clunky
> control shell script) and thus has a lot more moving parts. In
> addition, OpenSER suffers from horrifyingly inadequate documentation;
> the previous version (1.1?) was a lot better (in that at least what
> third-party documentation I could find matched the functionality).

Sad but true. I had to get some help from others to get mine setup, but
in the end they both fed from the same SQL database.

> If you're only doing onesie-twosie users, Asterisk seems like the way to
> go; if you're wanting to support, say, 20K users, OpenSER seems more
> appropriate.

I've had Asterisk fail to cope with 200 handsets all doing SIP
registrations, not even calls, on a 3Ghz P4 I think it was, stuck
OpenSER in front and used Asterisk for basically voicemenu/voicemail and
everything was much better.

> Oh, also though OpenSER might scale better, my experience is that
> Asterisk more fully embraces the concept of "graceful degradation"--
> when Asterisk gets overloaded, it gets slow, but behaves sanely and goes
> back to normal operation when the load passes; OpenSER seems to
> sometimes lose its mind when it gets overloaded and only a restart
> (perhaps with an associated wipe of its mysql tables) will bring it back.

Sounds like you hit a bug or a config issue to me, same hardware as
above was coping with hundreds if not thousands of simultaneous calls
and no load issues.

I know of one VSP (VoIP Service Provider) that more or less dumped
asterisk cause of all the load on their systems, much to the abhorrer of
their user base because IAX support was going away. He hasn't regretted
moving away from primarily using Asterisk as the number of concurrent
calls he can handle went through the roof, without buying any more hardware.

Also Asterisk can't cope with multi-threading so most people needing to
run Asterisk beyond a single core usually end up cheating and sticking
multiple copies of Asterisk running load balanced in front by OpenSER :)

--

Best regards,
Duane

http://www.freeauth.org - Enterprise Two Factor Authentication
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://e164.org - Because e164.arpa is a tax on VoIP

"In the long run the pessimist may be proved right,
but the optimist has a better time on the trip."
Any comments?!
 

cowboy47

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#6
My point is stil the same. You would be hard pressed to get equipment that will support that type of density. Just look at your interface card abilities. You could use Sangoma cards which can have 4 E1s on the card and then you can chain additional cards, but you are going to have space issues, slot issues, timing issues & power issues.

There is a module for Centos to run with more than 2GB of memory and there is references to that in other threads. Again, I think it would be foolish to dump Asterisk for OpenSer for the density issues. I also question this statement of putting OpenSer in front of Asterisk and things we wonderful. If the documentation is sparse and you have to look for help, and if you have to look for help for asterisk and the user base is bigger; if you run into problems with the other you will just be SOL.

C
 

zenny

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#7
I am still a bit divided. I wish I have had a look at some technical comparisons in reference to standalone asterisk performance against the openser (either standalone or as a SIP proxy to asterisk).

However, you (cowboy47) is right to state the fact that I can find more support for asterisk than for openSER. I appreciate that.

If anyone has any pointers, please let me know. Thanks!<br><br>Post edited by: zenny, at: 2008/02/17 20:24
 

DStirrup

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#8
I would be absolutely fascinated to see a PPT saying Asterisk can handle thousands of call in a single server. Seeing it in action would be more delightful. Maybe with 1.6 and a total core rewrite for SMP architecture...

Short of it all is that the SIP stack in Asterisk is messy, probably as a result of evolution. Look thru the digium bugs and you will see many comments from Dig staff. When confronted over SIP issues they recommend chatting about the issue after you have tested against current SVN 1.4.x Reason is they corrected structural issues during the upgrade to 1.4 (their comment to me about a SIP state bug).

But Cowboy you seem to be missing the point of SER. This is not an issue of how many trunks you have, it is an issue of how many SIP registrations you have! I have played with it and it is a royal PITA to setup so in small implementations it is hardly worth the effort excepting where there are many SIP hints aka BLF. What I did play with was having SER front ending Asterisk which worked well.

My understanding where OpenSER is used to best advantage is when you have more than maybe 200 SIP extensions. An example is that given above is where you have a VoIP trunk provider with maybe 10,000 subscribers and some distributed media gateways providing say 8 or 16 E1's (520 channels) attached to the PSTN where required. In that environment OpenSIP will shine.

The other condition I mentioned above is where you have many SIP Hints operating. For example if you have a few department secretaries who monitor each sections extensions in a console. Say 50 staff in each department, each user handset monitors maybe 6 other xtns so 300 hints + the section secretary adds another 50. I can tell you this really loads Asterisk down. If this was a 200 xtns site then 200 SIP registrations and maybe 1400 SIP Hints flying around makes for one very busy Asterisk box. Oh and you wanted a team leaders conference call with the big boss ....Oops sorry about the crash guys we'll have the phone system running again in a couple of minutes..

The SIP scalability issue is real today. Recent changes to the 1.4 code addressed important issues like preserving state of SIP Hints during Asterisk reloads. Unfortunately that slowed things down a fair bit when you have lots of them. Will Asterisk 1.6 fix these issue up? I hope so.

I have worked with a friend who runs/owns a couple of 80 seat asterisk call centres which have 4 -5 supervisors monitoring realtime what is going on with every operator. The system is so well managed that he has a display board showing realtime profitability for all staff to see. Point for him is that 120 simultaneous calls is about the limit of what a single asterisk box can do stabily without the customised code Digium use in their business editions. This is why you pay Digium monie$.
 

DStirrup

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#9
Why does Joomla post twice??? Oh well.<br><br>Post edited by: DStirrup, at: 2008/03/07 16:58
 

cowboy47

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#10
I understand your point. My questions are what are the other factors that exist that may be affecting this limitation? While I think it is great to have this information, I am still left wondering. What are the configurations of the server(s), is there not a design issue here where this limitation could be overcome.

Understand, I am not making light of the fact that digium may not give all the insight that they have on issues, however, if you go completely over to Openser, what are you opening yourself up to?

Many years ago, when I worked for Cisco, a Director made this comment to me about our competition. He said: "We all have warts. The thing is, the customers know where are warts are and can live with them." So, what are OpenSer's warts?

C
 

DStirrup

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#11
SER's warts are that it is trying to sort of be an Asterisk.. just joking sort of. Real warts would have to be how small the developer team and user support base is. Even when they finally finish it are they going to be too late to have any impact of the technology set?

I suppose my thoughts are that it is a great SIP switch which is trying to become a PBX whereas Asterix was always a PBX that awkwardly grew up. I think SER will get there IF someone develops a decent GUI application system like FreePBX. No I am not saying FreePBX is the greatest but it is a recognised starting point.

To be honest there are a few good switches out there but few have any decent configuration management or application systems. I personally believe if it wasn't for AMP there would never have been the excitemnt over Asterisk we currently see.

Until the OpenSER crew reach their design goals I will only occasionally check it out as I have too many other things to do.

I certainly would not implement a SER only project for my typical clients. But using it to bufer Asterisk, yes I like that.
 

cowboy47

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#12
OK, well stated. But here are some additional questions and comments I have. I have not read all the docs yet but I did not come across anything on the web site which would suggest the hardware configuration or supported peripherals. I did note that they seem to prefer Debian or SUSE linux, while not a big deal, is an issue of the install. (Laziness on my part because there is no iso that I found). SUSE is not free so that is another item. They do seem to offer support for Solaris (which if I were going to have to pay for somethng, I would rather do that).

Also, you are suggesting using OpenSer as a buffer or gateway to other asterisk systems. What are the design guides for that? What are the suggestions or limitations?

C
 

vitsoft

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#13

DStirrup

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#14
Thanks for that, great data.

Begs the qustion to me why the comparative slow down for my systems. The answer I think is FreePBX. It has a serious overhead in it's call /dialplan flow logic if you look at the logfiles compared to a straight forward handcrafted configs..

So if I dust off my lazy brain and start writing my own configs then maybe 1.4.x won't be so bad...<br><br>Post edited by: DStirrup, at: 2008/03/10 17:47
 

vitsoft

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#15
Ok.
If you have a situation in which the asterisk machine is very stressed and loaded remember that a realtime configuration coul help.
Unfortunatlly freepbx don't support relatime.
 

cowboy47

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#16
OK, I have a question for you. After all these conversations, I attempted to install openser. What a nightmare. At first, after all the references to the system wanting Debian, I installed Debian 4 on a system and found that certain libraries that are required for Openser do not exist or can't be installed. So then I went to CentOS 5.1 thinking that would have the latest. Same thing, I can get to a certain point but cannot compile and I am not sure about setting the environmental variables.

Have you installed this?

C
 

vitsoft

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#17
I use opensuse 10.3 and you can find the rpm of openser on the network/telephony repository.

But this is only the sw ... then you have to configure it !!
 

rdsousa

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#18

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