Re: [Freeswitch-users] mod_conference scalability
Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like a configuration error. If not, I already see the title of the next Digium blog entry: FreeSwitch scalability myth finally ends: The worst Asterisk version ever (1.4) beating the crap of the best and latest FS. Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins the final conference battle! :-) François. On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 3:49 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability We didn't post it anywhere but we just get overwhelmed with them and many of them are unfounded and take up a lot of time to track down. That does not mean you have not found a real problem but the first step is trying trunk. On Thu, Dec 17, 2009 at 2:32 PM, Brian br...@proximosystems.com wrote: I didn’t realize there was a policy about load testing questions. What forum should I have used for this? I didn’t get the chance to test on FS trunk yet, but when I do I will provide you with the feedback when I do. Just let me know what forum to use for this topic from now on. Thanks, Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 2:42 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability One man's stable release is another man's 6 month old release with hundreds of known fixed bugs. If one of the core developers tells you to try it, you may as well take the time to try it now that you have opened a forum questioning the scalability. When you tested asterisk did you actually use 600 phones and verify that each one can hear the audio perfectly and in time with what the speaker was saying? Did you try same on FS? Did you optimize your dialplan on FS to deal with a load test or follow any of the recommended performance tuning page. All of the answers to these questions are really moot because we have a policy against entertaining load testing questions but if you like asterisk, by all means, use it, and good luck to you if those numbers you are testing at are what you plan to put in real production. On Thu, Dec 17, 2009 at 1:29 PM, Brian br...@proximosystems.com wrote: Hi Mike, I didn’t get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able to get 1 speaker and 600 listeners on a single conference with no audio issues. The CPU at that point was just over 300%, same as where the single conference scenario failed on FreeSWITCH with 300 listeners. I was able to push it to over 700 listeners before I reached 400% CPU usage (I guess maxing out my quad-core processors), and asterisk finally crashed. But up until that point, there were no audio problems. I’ve read a lot about how FreeSWITCH is supposed to be more scalable than Asterisk, but unless there is something wrong with my FreeSWITCH setup, Asterisk was clearly the winner in this test – more than doubling FreeSWITCH capacity in this case. Again, maybe there is something on the FreeSWITCH side that I’m doing wrong, but I don’t see what it could be. Brian. From: Michael Jerris [mailto:m...@jerris.com] Sent: Thursday, December 17, 2009 10:18 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability I would be curious what the same tests produce with svn trunk of FreeSWITCH. Mike On Dec 16, 2009, at 4:49 PM, Brian wrote: Hi, I’m new to FreeSWITCH and I’m testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn’t scale nearly as well as I’d hoped (based on what I’ve read on how FreeSWITCH is supposed to be generally very scalable). Here’s my server
Re: [Freeswitch-users] mod_conference scalability
It was of course just bad humor, I love both projects for what they are, and I agree that both have their own advantages and inconvenients. For example, accessing that same conference from a dahdi card could be another goal where Asterisk would be at an advantage, as chan_dahdi is still superior (in the more tested sense) than openzap+mod_openzap. I just use both projects separately or together depending on what's needed! I'm no banker nor do I understand the code, but many thanks for all those unpaid contributions providing an excellent alternative for free telephony. Your names really deserve being engraved in google's cache for eternity. :-) But still, I would like to see those numbers... François. On Fri, 2009-12-18 at 10:34 -0600, Anthony Minessale wrote: Conferencing is hardly the best place to judge performance. Quality is a far more important goal to me in conferencing. Lets compare who can do 48khz conferences with several 32k siren callers on a polycom 6000, several more using G722 at 16khz and another handful of people on g711 ulaw all at different rates and ptimes talking in near-real time with low delay and low echo. The fact that you can broadcast the conferences to icecast, control it from an external application and play files etc, and oh yeah, it can stream video. Frankly, considering this is a free software project and so many people benefit, i would rather focus on quality than what numbers i can get from having robots call the conference in some way that probably does not match reality. I would love for someone to sponsor the effort to add features to the conference module, but of course, I do not hold my breath, instead I continue to improve it for free when I find time. This is one of many reasons I do not enjoy performance discussions unless I am talking to an engineer who understands the code or a banker ready to pay for improvements. That is not my way of saying pay me or forget it as you can clearly see the conference module has made it to where it is today with no financial support at all. Just the efforts of myself and several brave volunteers over the years who have contributed to it. BTW, We have a weekly call, there is one today in 30 minutes. Drop by sip:8...@conference.freeswitch.org This is just an openVZ instance mind you running at 48khz waiting for anyone to call in and say hi. On Fri, Dec 18, 2009 at 10:12 AM, François Delawarde fdelawa...@wirelessmundi.com wrote: Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like a configuration error. If not, I already see the title of the next Digium blog entry: FreeSwitch scalability myth finally ends: The worst Asterisk version ever (1.4) beating the crap of the best and latest FS. Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins the final conference battle! :-) François. On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 3:49 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability We didn't post it anywhere but we just get overwhelmed with them and many of them are unfounded and take up a lot of time to track down. That does not mean you have not found a real problem but the first step is trying trunk. On Thu, Dec 17, 2009 at 2:32 PM, Brian br...@proximosystems.com wrote: I didn’t realize there was a policy about load testing questions. What forum should I have used for this? I didn’t get the chance to test on FS trunk yet, but when I do I will provide you with the feedback when I do. Just let me know what forum to use for this topic from now on. Thanks, Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 2:42 PM To: freeswitch-users@lists.freeswitch.org Subject: Re
[Freeswitch-users] sofia gateways and linux multipath routing
Hello all, I'm interested in using mod_sofia with multiple Internet connections (configured as a unique load-balancing route using multipath). One solution would be to define a different profile for each connection, but it would be more practical having a unique external profile that would automatically handle everything (detecting multiple public IPs and selecting the right one for a call, being able to select the router for gateway registration...). Routing table: 192.168.10.0/24 dev eth0 proto kernel scope link src 192.168.10.1 192.168.1.0/24 dev eth1 proto kernel scope link src 192.168.1.2 192.168.2.0/24 dev eth2 proto kernel scope link src 192.168.2.2 default proto static nexthop via 192.168.1.1 dev eth1 weight 1 nexthop via 192.168.2.1 dev eth2 weight 1 Both default routers (192.168.1.1 and 192.168.2.1) would have a distinct public IP. Several questions cross my mind: - Can a unique sofia profile be bound to multiple IPs (not 0.0.0.0)? - How would FS behave with a unique external profile in that situation? * Would FS reply to an incoming call using the same router it came from forcing packet source address? * Would FS stick to a unique router for all flows of an outgoing call (SIP, RTP, UDPTL)? * Can I force a gateway to use a given router (for calls, registration, ...)? * Would the NAT system (using stun or auto-nat) work in that situation, or does it assume only one default router (and a unique public IP) exists per profile? - Knowing the above, would it be necessary to use a different profile for each router/interface, and define the same gateway in each of these? - Tricky question: What if multiple routers are on the same network/interface (192.168.1.1, 192.168.1.2, ...)? Thanks in advance, François. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Any advances on T.38 support for FS?
Is there any work planned for T.38 termination (in mod_fax)? François. On Tue, 2009-06-30 at 12:07 -0400, Michael Jerris wrote: We currently support t.38 passthrough only using proxy_media mode. T. 38 gateway is on the roadmap but not yet close to complete. Mike On Jun 30, 2009, at 5:15 AM, François Delawarde wrote: Many issues on Asterisk's T.38 (or probably just on T.38?)... Could it convince those relying on this modern version of a 50yo technology to switch to and with FreeSwitch? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Any advances on T.38 support for FS?
Great news! For what I could read, the most famous DSP programmer worldwide (Steve) seems to be helping out for mod_fax. I guess I should register to freeswitch-dev to monitor this closely. Thanks, François. On Wed, 2009-07-01 at 18:38 +1000, Jason White wrote: François Delawarde fdelawa...@wirelessmundi.com wrote: Is there any work planned for T.38 termination (in mod_fax)? Yes, as discussed on the mailing list recently. If you're volunteering to help, I'm sure the FreeSWITCH developers would appreciate contributions of code. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Any advances on T.38 support for FS?
Many issues on Asterisk's T.38 (or probably just on T.38?)... Could it convince those relying on this modern version of a 50yo technology to switch to and with FreeSwitch? François. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Sounds order
Hi Brian, Is FreeSWITCH going to have Spanish/French/... sounds as well or do those need to be? Thanks, François. On Tue, 2009-05-26 at 11:36 -0500, Brian West wrote: I'm getting ready for the next sound file order for FreeSWITCH. I have a rather large set of files to be recorded for the zRTP integration if anyone wants to help out. ;) Please contact me off list. I would like everyone to update and try out voicemail and nitpick anything that you feel is wrong there too and let me know so I can have them corrected in this order also. Thanks, Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] openzap and progress detection
I briefly looked at the code, so I will then answer my own questions from what i understood (and I didn't understand much). Please correct me if I'm wrong (see inline answers). A new question would then be: Are any of those features planned for the future of OpenZAP? On Mon, 2009-05-18 at 10:29 +0200, Francois Delawarde wrote: Hello, I'm in Spain with an analog TDM400 Clone from OpenVox (with 1xFXO +1xFXS), and am trying first to make the FXO work with Openzap and Freeswitch (using dahdi 2.2.0-rc4). Openzap perfectly detects and loads the spans, but I'm currently enable to dial out with the FXO module, it doesn't dial anything and times-out after 30 seconds. I believe it has to do with some tone detection and therefore have a few questions: - When I plug in the line, dahdi sends an event (event 17) that is ignored by OZ. Can we enable some type of battery check in OZ before dialing out, or is there some variable to monitor battery (oz dump doesn't show the battery status)? No. - Does OZ use the polarity switch events sent by dahdi (in kewlstart mode) for answer and hangup detection? Yes. - Apparently, OZ does tone progress detection by frequency, but here in Spain, most tones use the same 425Hz frequency with different on-off timing. Is it possible to detect those? No. - As some PBX in Spain transfer calls by first hanging up and picking up on another phone, can we enable/disable parts of polarity switch and/or tone progress detection (ex: (hangup)/(answer)onpolarityswitch in Asterisk) No. - Some lines here are connected to very old FXS from operators that have low sound quality and can take a few seconds to give a dial tone when picking up. Is it possible to introduce a delay before sending DTMF digits when dialing? Is it possible to relax DTMF detection, and tweak DTMF settings (make them a bit longer, with a longer pause for the other side to detect)? No / No idea / No idea My ideal case to make it work in every case around here would be to: - have OZ fail to dial if battery is not present (and be able to fetch battery status somehow) - disable tone progress (sometimes call ends up on some local PBX that answers and provides US tones which are different) - be able to have an initial pause before dialing with DTMF digits - use polarity switch to detect remote answer, but not hangup (for transfer issues) Is the above possible? No. Thanks in advance, François. You're welcome! François. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] openzap and progress detection
Ok, feature request for progress detection opened as OPENZAP-70. Meanwhile, I humbly offer a basic working environment in Spain (a TDM400 with 1xFXO) if you need it for testing. François. On Wed, 2009-05-20 at 08:08 -0500, Anthony Minessale wrote: The most difficult one would be the cadence detection. The rest are just based on events from the driver. We have never tried it in Spain so we unfortunately have not had a working environment to test it in. I am sure we could strive to support your requests but it will take time and resources so maybe you can open a feature request on jira http://jira.freeswitch.org and we can try to keep track of your endeavor. On Wed, May 20, 2009 at 5:26 AM, François Delawarde fdelawa...@wirelessmundi.com wrote: I briefly looked at the code, so I will then answer my own questions from what i understood (and I didn't understand much). Please correct me if I'm wrong (see inline answers). A new question would then be: Are any of those features planned for the future of OpenZAP? On Mon, 2009-05-18 at 10:29 +0200, Francois Delawarde wrote: Hello, I'm in Spain with an analog TDM400 Clone from OpenVox (with 1xFXO+1xFXS), and am trying first to make the FXO work with Openzap and Freeswitch (using dahdi 2.2.0-rc4). Openzap perfectly detects and loads the spans, but I'm currently enable to dial out with the FXO module, it doesn't dial anything and times-out after 30 seconds. I believe it has to do with some tone detection and therefore have a few questions: - When I plug in the line, dahdi sends an event (event 17) that is ignored by OZ. Can we enable some type of battery check in OZ before dialing out, or is there some variable to monitor battery (oz dump doesn't show the battery status)? No. - Does OZ use the polarity switch events sent by dahdi (in kewlstart mode) for answer and hangup detection? Yes. - Apparently, OZ does tone progress detection by frequency, but here in Spain, most tones use the same 425Hz frequency with different on-off timing. Is it possible to detect those? No. - As some PBX in Spain transfer calls by first hanging up and picking up on another phone, can we enable/disable parts of polarity switch and/or tone progress detection (ex: (hangup)/(answer)onpolarityswitch in Asterisk) No. - Some lines here are connected to very old FXS from operators that have low sound quality and can take a few seconds to give a dial tone when picking up. Is it possible to introduce a delay before sending DTMF digits when dialing? Is it possible to relax DTMF detection, and tweak DTMF settings (make them a bit longer, with a longer pause for the other side to detect)? No / No idea / No idea My ideal case to make it work in every case around here would be to: - have OZ fail to dial if battery is not present (and be able to fetch battery status somehow) - disable tone progress (sometimes call ends up on some local PBX that answers and provides US tones which are different) - be able to have an initial pause before dialing with DTMF digits - use polarity switch to detect remote answer, but not hangup (for transfer issues) Is the above possible? No. Thanks in advance, François. You're welcome! François. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo
Re: [Freeswitch-users] any way ring fifo members one by one?
Hello,
Anthony, I would like to provide a patch allowing having different call
distribution strategies, at least for call back agents.
Do you think the simple approach of modifying the SQL query in
find_consumers (given strategy that would be set from dialplan) would be
enough?
Thanks,
François.
On Mon, 2009-05-04 at 08:25 -0500, Anthony Minessale wrote:
On Sun, May 3, 2009 at 11:01 PM, seven dujinf...@gmail.com wrote:
Actually, for the call back agents, because the fifo use
originate to start a new session, the new session won't hang
up unless one agent answered or timeout. Agents will hear
nothing and wait(member_wait=wait) on the queue or
hanup(nowait) if caller hang up before an agent answer the
phone. '
When you are using on-hook agents, it's presumed to be under low call
volume, you can just set the agents to get popped
into the queue in nowait mode so if the caller changed his mind the
agent will get a hangup. Remember, if there are X customers in the
queue, mod_fifo generates X outbound calls to try to service them.
And I also found out the the member timeout doesn't work but
call_timeout works in a dial string. Is it a bug I should
reported to jira?
fifo name=sales_f...@$${domain} importance=0
member timeout=10 simo=1
lag=5{call_timeout=6,fifo_member_wait=nowait}user/1009@
$${domain}/member
/fifo
call_timeout is only valid on inbound legs to set the timeout it's
willing to wait for a caller to answer. You are confusing it with
leg_timeout which is designed to go in the {}
And even the timeout works, it's not ideal. It's better to
bridge to an agent other than originate I think. Keep looking.
I am not sure what you mean by that. bridge instead of originate?
The process is to originate the call and then bridge the agent to the
caller. All calls in FS start out as origiante
If you want app_queue you are welcome to download and use it from
http://www.asterisk.org
On Apr 29, 2009, at 4:27 PM, François Delawarde wrote:
Hi,
It should be easy to modify mod_fifo to include this
functionality.
Correct me if I'm wrong:
For call back agents at least, when X calls are in the the
queue, Freeswitch tries to search for up to X agents in
database. This algorithm is much more optimized than
Asterisk, as Asterisk will take calls one by one and try to
connect them to an agent, it should then stay as it is.
The simplest idea to control the call distribution algorithm
would be to modify the database query in the
find_consumers function (right now, the algorithm is:
order by outbound_call_count). A variable could control
the order by of this query, and the problem would be
solved at least for call back agents. I guess sqlite3
should allow very complex queries, but I don't know if there
could be performance issues.
Do you think it is a possible -trivial- solution?
François.
On Wed, 2009-04-29 at 08:46 +0200, Antonio Gallo wrote:
seven ha scritto:
oh, thank you Antonio. I think it would be better to collect
more
ideas before open a bounty. And I more interested in
playing(including
patching the code) with that than use the function.
I was working on other stuff yesterday and just looked at the
wiki:
- it seems there is already a bounty for something like that;
- there is a wiki page about how to implement it with Javascript,
ofc
you need to tailor it to your own needs;
AgX
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Re: [Freeswitch-users] any way ring fifo members one by one?
Hi, It should be easy to modify mod_fifo to include this functionality. Correct me if I'm wrong: For call back agents at least, when X calls are in the the queue, Freeswitch tries to search for up to X agents in database. This algorithm is much more optimized than Asterisk, as Asterisk will take calls one by one and try to connect them to an agent, it should then stay as it is. The simplest idea to control the call distribution algorithm would be to modify the database query in the find_consumers function (right now, the algorithm is: order by outbound_call_count). A variable could control the order by of this query, and the problem would be solved at least for call back agents. I guess sqlite3 should allow very complex queries, but I don't know if there could be performance issues. Do you think it is a possible -trivial- solution? François. On Wed, 2009-04-29 at 08:46 +0200, Antonio Gallo wrote: seven ha scritto: oh, thank you Antonio. I think it would be better to collect more ideas before open a bounty. And I more interested in playing(including patching the code) with that than use the function. I was working on other stuff yesterday and just looked at the wiki: - it seems there is already a bounty for something like that; - there is a wiki page about how to implement it with Javascript, ofc you need to tailor it to your own needs; AgX ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
