[OpenSIPS-Users] Freeswitch vs Asterisk

Fri Dec 10 16:11:26 CET 2010

Guys,

Point taken.  Personally I prefer Coke over Pepsi.

- Opensips user Jeff

On 12/10/10 10:04 AM, "paul.gore.j at gmail.com" <paul.gore.j at gmail.com>
wrote:

>I haven't seen many posts from frustrated peole, majority of them come
>from people either selling fs based services or part of fs development
>team.
>From my experience with fs 1.0.4 it was crashing every 2 months, 1.0.6 is
>better, I already posted crashing rate for our use case.
>I haven't experienced any stabilty issues with * 1.6 yet, but it only
>sees light traffic.
>FS is a great piece of software but it does have issues, sometimes even
>simplest things like "find me" function work flawlessly in * and pain in
>the ass to impelement in fs due to either bad nat handling or some other
>bugs.
>
>
>-----Original Message-----
>From: Erik Dekkers
>Sent:  12/10/2010 3:28:11 AM
>To: 'paul.gore.j at gmail.com'; 'OpenSIPS users
> mailling list'
>Subject:  RE: [OpenSIPS-Users] Freeswitch vs Asterisk
>
>The reason people are yelling on the internet "Freeswitch is much better
>than asterisk" is pure frustration.
>They have used asterisk for years, were faced with crashes and since they
>are using freeswitch they don't see those crashes anymore (apart from the
>reason of those crashes).
>No wonder they tell everyone freeswitch is better than asterisk. From
>their point of view asterisk is bad.
>
>It's not Mr. Collins opinion that asterisk is worse than freeswitch. It
>are the ex-asterisk people who are saying that, think about that.
>
>-----Oorspronkelijk bericht-----
>Van: users-bounces at lists.opensips.org
>[mailto:users-bounces at lists.opensips.org] Namens paul.gore.j at gmail.com
>Verzonden: donderdag 9 december 2010 16:27
>Aan: OpenSIPS users mailling list
>Onderwerp: Re: [OpenSIPS-Users] Freeswitch vs Asterisk
>
>I just want to reply to mr Collins with FS: your post looks very much
>like advertisement, and I have seen that "fs is so much better than *"
>all over internet from people connected to fs. That is unethical to say
>the least.
>In fact we have exprerienced fs crashes with core dump at least  once in
>6 months and we process just under 40K calls/month.
>As to "nat tools" which you mentioned they just do not work. In fact
>usually * box works much better for natted users.
>As to xml curl interface - we do use it, and it's a pathetic way to feed
>a dialplan to a switch, since it's inefficient resource wise, but there
>was no other way available for real time solution where's * supports real
>time db out of the box.
>Trust me we do have development experience with both * socket interface
>and fs one, and in my opinion * solution is far better and has far less
>bugs.
>
>-----Original Message-----
>From: James Mbuthia
>Sent:  12/08/2010 5:55:42 PM
>Subject:  Re: [OpenSIPS-Users] Freeswitch vs Asterisk
>
>From the comments mentioned it seems FS meets my core requirements which
>are scalability and stability. I don't have the financial and manpower
>resources for a large scale implementation so am looking at getting a
>high end server and a solution that can scale well until I can through in
>more resources. It seems also FS is more stable than * which is a huge
>plus for a small operation like mine and since I only need few features
>from the solutions available then FS makes more sense
>
>On Wed, Dec 8, 2010 at 8:29 PM, Michael Collins <msc at freeswitch.org>
>wrote:
>
>> Dave,
>>
>> Thanks for your two cents. :)
>>
>> Regarding the PRI stuff, Sangoma is really doing a lot with FreeTDM
>> (the replacement for OpenZAP) and it will be a full-featured PRI
>> stack. If you're missing anything in the PRI implementation then
>> Moises Silva would definitely want to hear about it.
>>
>> On the voicemail stuff we have heard similar reports. In fact, we have
>> an intrepid community member who is building "Jester Mail" as a FS
>> alternative to Asterisk's Comedian mail. The basic idea is that Jester
>> Mail will be 100% customizable such that you can drop in FS as a
>> replacement for Asterisk and your voicemail users would be none the
>>wiser.
>>
>> By early next year you will probably have more options if you wish to
>> swap out your remaining Asterisk servers.
>>
>> -MC
>>
>>
>> On Wed, Dec 8, 2010 at 9:53 AM, Dave Singer
>><dave.singer at wideideas.com>wrote:
>>
>>> We have both asterisk and Freeswitch in production. The primary place
>>> where we have * installed is as a pbx for our business customers
>>> (where we started doing business and didn't know any better). We are
>>> still using * for them for two reasons: migration time and voicemail
>>> app I feel is still better in a couple points. They are low volume
>>> usage so crashes are very rare.
>>> We also have some boxes where we connect to telecom PRI circuits
>>> where the API for FS doesn't support some params we need to set. So
>>> we are stuck there for now. There systems handle moderate volume, 30 -
>>>90 simultaneous calls.
>>> This call volume has proved to be deadly to asterisk and we have to
>>> restart asterisk daily or suffer a crash in the middle of peek times.
>>> We use FreeSwitch as the workhorse with a custom routing module
>>> combined with Opensips as a class 4 switch (whole sale trunking
>>> service). With high powered servers (latest dual xeon quad core, 16GB
>>> ram, and 10Gbit ethernet) it can handle thousands of simultaneous
>>> calls. They run for months without problem (would be longer but for
>>> reboots for upgrades, etc., not FS crashes).
>>> We also have a class 5 system that handles residential users which
>>> uses FS and opensips for failover. Again no FS crashes.
>>> FS is also our conference server for all our services.
>>>
>>> We started out using * building the business PBXs. Later found FS as
>>> we were developing the residential system and converted to using it.
>>> Coming from * to FS has some difficulties because of the different
>>> ways of doing things like the flow of the dialplan where all
>>> conditions are evaluated at the time of entry to the dialplan, not as
>>> each line is executed (executing another extension solved this problem
>>>for me).
>>> I do think FS has a little higher learning curve, I have found it
>>> better in almost every area, especially stability and flexibility.
>>>
>>> Well, those are my 2 cents. :-D
>>> Dave
>>>
>>> On Tue, Dec 7, 2010 at 11:27 AM, Michael Collins
>>><msc at freeswitch.org>wrote:
>>>
>>>> Comments inline. (Full disclosure: I am on the FreeSWITCH team, so
>>>> if I come off as biased then you know why. ;)
>>>>
>>>> On Tue, Dec 7, 2010 at 8:29 AM, paul.gore.j at gmail.com <
>>>> paul.gore.j at gmail.com> wrote:
>>>>
>>>>> We use freeswitch in prod alone, no opensips yet. I would say fs is
>>>>> definetly more scalable than *.
>>>>> Stability wise seems like fs is on par with *.
>>>>>
>>>> YMMV, but a large percentage of FreeSWITCH users have abandoned
>>>> Asterisk specifically because of stability issues, like random and
>>>> inexplicable crashes.
>>>>
>>>>
>>>>> * has substantially better interface for control over socket
>>>>> connection
>>>>> - it's easier to implement and it's more consistent.
>>>>>
>>>> This statement is patently false. The FreeSWITCH event socket
>>>> interface is incredibly powerful and is absolutely more consistent
>>>> than the AMI. Those wondering about inconsistencies in the AMI
>>>> should listen to a seasoned AMI developer talk about the challenges:
>>>> http://www.viddler.com/explore/cluecon/videos/29/
>>>>
>>>>
>>>>> Configuration wise, I think * is easier, xml- based approach in fs
>>>>> is cumbersome and has no real advantage over *.
>>>>>
>>>> This one really is like Coke vs. Pepsi. Some people hate XML, some
>>>> people hate INI-style config files. Personally, I've done both and
>>>> now that I'm accustomed to FreeSWITCH's XML files I find them much
>>>> easier to read than Asterisk's config files. There is one "real
>>>> advantage" to using XML for configs and that is that machines and
>>>> humans can both produce XML, so it's relatively simple to let a
>>>>machine generate XML-based configs on the fly.
>>>> (FreeSWITCH uses "mod_xml_curl" as the basis for dynamic
>>>> configuration - it's very cool and I recommend that you check it
>>>> out.)
>>>>
>>>>
>>>>> We have endless problems with fs nat handling, lots of no audio
>>>>> issues with end users behind a nat. That's why we want to try
>>>>> opensips solution for that.
>>>>>
>>>> Almost all NAT problems stem from phones which don't handle NAT
>>>> properly or NAT devices that scramble ports and IP addresses when
>>>> packets pass through. FreeSWITCH has several NAT-busting tools to
>>>> assist the system admin. Some tools are for when FS is behind NAT,
>>>> others are for when the phones are behind NAT. Bottom line is this:
>>>> if the NAT device and the phones are not horribly broken then FS
>>>> works great with NAT and in many cases "just works." However, when
>>>> you start mixing crazy scenarios with broken phones then bad things
>>>> will happen. Example: Polycom phones are wonderful except that they
>>>> don't support rport - FS has a mechanism to assist with this but if
>>>> you turn it on to "fix" the Polycom phones then it will break all
>>>> other phone types. (There is a limit to the amount of pandering that
>>>> the FS devs will do in order to interop with broken devices. In many
>>>> cases they simply say "NO" to doing stupid things in order to work
>>>> with broken devices. If you must work with such a device then
>>>> perhaps FreeSWITCH isn't for you.)
>>>>
>>>> All that being said, the FreeSWITCH developers have a simple mantra
>>>> that they follow to the letter: Use what works for your situation.
>>>> If Asterisk works for you then by all means use it! You won't hurt
>>>> our feelings. (I work daily with the FreeSWITCH dev team.) If you
>>>> have people knowledgeable in Asterisk or FreeSWITCH then it might be
>>>> advantageous to go with the project for which you have more
>>>> resources. In any case, if you are interested in FreeSWITCH we have
>>>> a great IRC channel (#freeswitch on irc.freenode.net), an actively
>>>> mailing list, and a small but growing international community of
>>>>users. You are most welcome to join us to see what we're about.
>>>>
>>>> Happy VoIPing!
>>>> -Michael S Collins
>>>> IRC:mercutioviz
>>>>
>>>>
>>>>
>>>>>
>>>>>
>>>>> -----Original Message-----
>>>>> From: James Mbuthia
>>>>> Sent:  12/07/2010 8:54:51 AM
>>>>> Subject:  [OpenSIPS-Users] Freeswitch vs Asterisk
>>>>>
>>>>> Hi guys,
>>>>>
>>>>> I want to integrate my Opensips implementation with either Asterisk
>>>>> or Freeswitch to do the following functions
>>>>>
>>>>> - Act as a Media server
>>>>> - Connect to the PSTN
>>>>> - Act as a B2BUA
>>>>>
>>>>>
>>>>> There's been alot of hype about Freeswitch and I wanted to know
>>>>> from people who've integrated it to OpenSIPS how it compares to
>>>>> Asterisk especially in the case of installation and intergration,
>>>>> scalability and ease of maintenance.  Any info would be a huge help
>>>>>
>>>>> regards,
>>>>> james
>>>>>
>
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