- FreeSWITCH-users - FreeSWITCH Mailing Lists

Announcement: FreeSWITCH on PPC, FreeScale, POWER platforms
by Larry Morley 09 Sep '15

09 Sep '15

Hello, We're working on porting FS to the POWER architecture, from PPC G4 to POWER 8, with specific hardware and software platforms to be determined by the level of interest expressed in any given one. We've strongly advocated the use of PPC/POWER processors since early on, for both general purpose computing and in specific problem domains. The decision by IBM, Tyan, and other industry leaders to undertake the OpenPOWER initiative, combined with the recent availability of a POWER 8 system from Tyan [http://www.tyan.com/campaign/openpower/index.html] has reaffirmed our commitment to, and strengthened our belief in, the potential of this architecture, and how outstanding it is from any number of perspectives - including cost vs. benefit. Currently, we're testing a build of a port of the latest release of FreeSWITCH using G4 and G5 based systems capable of running Debian "Jessie". We've been careful to maintain the structure of the supporting libraries, building them only as .deb packages. Feature test results so far have been better than promising; and, we have no reason to believe the results of the remaining tests will be any less so. If you're interested in development of FS on a PPC / POWER platform, or would like more information on the current status of our project, please contact me. Thanks, Larry Larry Morley R&D Director Enlightened Computer Solutions, L.L.C. lmorley(a)neny.cslimits.net TEL: (442) 333 3416 FAX: (518) 284 0373

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Condition on if user exists.
by Michael Nielsen 09 Sep '15

09 Sep '15

I would like to the following: <condition IF-USER-EXISTS> </condition> <condition IF-USER-DO-NOT-EXISTS> </condition> I'm not able to check on regexp, as my user ID's match +44-12345678... And I would like to do one thing if the user exists and otherwise route to my PSTN gateway. I've tried to just use bridge with both local routing and PSTN calls, but then I try to call my PSTN everytime I call local users.

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Multiple Conditions
by Tim King 09 Sep '15

09 Sep '15

I need to route calls to different destinations based on network ACL. I still need the calls to match a certain dial pattern how can I accomplish this? *All calls will match this first condition* <condition field="destination_number" expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$"> *If this* <condition field="${acl(${network_addr} list1)}" expression="true"/> *execute these actions* *if this* <condition field="${acl(${network_addr} list2)}" expression="true"/> *execute these actions*

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Hardware Recommendations
by Adrian Thompson 09 Sep '15

09 Sep '15

Hello, We are new to FreeSwitch and would like to set up a PBX system with a BigBlueButton on the side. I've played around with BigBlueButton and have it working pretty flawlessly. - We would like to have two PBX (Fusion PBX GUI) systems at two sites with three remote sites piggy backing through VPN. - 10 Phone Numbers, 50 Extentions - Fail over to second site if first site is down - We have 4 Fax lines - We are looking for phones that can set presence (Available, Away, DnD, Custom) and have good quality sound - We should be able to support at least 3 lines per phone, but if a two line model with the above specs is available, we can work with that also I have looked over the hardware list on the FreeSwitch site, but I don't have too many resources to test different phones before implementation. The hardware I have looked at so far is the Polycom VVX 300 and The VVX 400 with expansion (Switchboard) but it looks like the DnD button doesn't really work as intended. What about Cisco Phones? I'd like to keep things under $100 USD for the IP Phone price per unit. SITE-1(Main)<---------->SITE-2(Fail-over) ^ | REMOTE-SITES-VPN Thank you in advance for any advice on the matter Adrian

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Hardware Recommendations
by Adrian Thompson 08 Sep '15

08 Sep '15

Hello, We are new to FreeSwitch and would like to set up a PBX system with a BigBlueButton on the side. I've played around with BigBlueButton and have it working pretty flawlessly. - We would like to have two PBX (Fusion PBX GUI) systems at two sites with three remote sites piggy backing through VPN. - 10 Phone Numbers, 50 Extentions - Fail over to second site if first site is down - We have 4 Fax lines - We are looking for phones that can set presence (Available, Away, DnD, Custom) and have good quality sound - We should be able to support at least 3 lines per phone, but if a two line model with the above specs is available, we can work with that also I have looked over the hardware list on the FreeSwitch site, but I don't have too many resources to test different phones before implementation. The hardware I have looked at so far is the Polycom VVX 300 and The VVX 400 with expansion (Switchboard) but it looks like the DnD button doesn't really work as intended. What about Cisco Phones? I'd like to keep things under $100 USD for the IP Phone price per unit. SITE-1(Main)<---------->SITE-2(Fail-over) ^ | REMOTE-SITES-VPN Thank you in advance for any advice on the matter Adrian

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Re: [Freeswitch-users] OneWay audio after hold->unhold
by kaleem rehman 08 Sep '15

08 Sep '15

Hi Prashanth, is there any reason you are going for SRTP? I always advise customers to first setup normal link and fully test then move to SRTP. when you are saying freeswitch not responding with key, are you seeing the message in FS Debug being generated? if not then you will need to raise a JIRA. if you can, then next step would be to see this in traceSM logs and see if SM100 is getting these and how is SM responding to these and if sending to CM or not. thanks, kaleem On Tue, Sep 8, 2015 at 11:08 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: OneWay audio after hold->unhold (Prashanth Devarajappa) > > > ---------- Forwarded message ---------- > From: Prashanth Devarajappa <Prashanth.Devarajappa(a)enghouse.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Sep 2015 10:07:27 +0000 > Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold > > Thanks for your message. > > > > >> Best Route to do this is via session Manager as this is the supported > method. First create a trunk link between Session manager and Freeswitch > > > > This is how we have setup our environment and got this far. > > > > Prashanth > > > > > > *From:* kaleem rehman [mailto:k4kaleem@gmail.com] > *Sent:* 08 September 2015 10:46 > *To:* freeswitch-users(a)lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 111, Issue > 82 > > > > Hi Prashanth / Bob, > > > > Avaya CM supports sip but this is mainly used for Aura VM, Experience > portal and SM and more of Avaya only devices, direct conenctions to > Asterisk , FS & 3CX work but I always experienced issues due to Avaya SIP > Limitations and also officially this is not supported > > Best Route to do this is via session Manager as this is the supported > method. > > First create a trunk link between Session manager and Freeswitch, then > register extension directly to SM and make calls between from this > extension and Freeswitch, once everything is working then create a trunk > between CM & SM, or use existing and setup digit routing on SM and CM. > > > > if you have any issues setting up SM and its entities then let me know. > this way will be best and far most easy one. > > > > thanks, kaleem > > > > On Tue, Sep 8, 2015 at 9:54 AM, < > freeswitch-users-request(a)lists.freeswitch.org> wrote: > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: OneWay audio after hold->unhold (Bote Man) > 2. Re: Compiling under SmartOS (Stanislav Sinyagin) > 3. Re: OneWay audio after hold->unhold (Prashanth Devarajappa) > > > ---------- Forwarded message ---------- > From: Bote Man <bote_radio(a)botecomm.com> > To: "'FreeSWITCH Users Help'" <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 7 Sep 2015 18:03:17 -0400 > Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold > > Actually, yes I need all of the settings on the Avaya side. > > > > I am not using encryption and I never could get the Avaya CS1000M to talk > to the FreeSWITCH installation. I think the closest I could get was the > signaling would allow the call to be answered, but I got no audio to pass > either direction. > > > > I think the Avaya puts information in unexpected elements of the SIP > message and I had no more time to troubleshoot it because it was not my > primary task. > > > > There must be some small setting that I am missing and I would love to > find the missing piece of the puzzle. > > > > Thanks! > > > > Bote > > > > > > *From:* Prashanth Devarajappa > *Sent:* Monday, 07 September, 2015 09:47 > *Subject:* Re: [Freeswitch-users] OneWay audio after hold->unhold > > > > *Avaya PBX Version Control* > > § Media encryption over IP:Y > > § Session Manager Element Manager 6.3 > > § System Manager Release 6.3.5 > > § Communication System Management 6.3.10 > > > > *In IP-Network-region :* > > § Codec set matches group for encryption > > § Intra-region IP-IP Direct Audio: yes > > § Inter-region IP-IP Direct Audio: yes > > > > *In CM, signaling-group of type*: *SIP* > > § Transport Method: TLS > > > > Let me know if you need further Avaya config details. > > > > *FreeSwitch :* > > Its standard FS with custom (code/config) changes to enable > > > > · Support for RFC 5939 : Capability Negotiation > > · Support for UNENCRYPTED_SRTCP > > · Support IP Shuffling with in ACK…… to be done J > > > > > > Prashanth > > > > > > > > *From:* Bote Man [mailto:bote_radio@botecomm.com <bote_radio(a)botecomm.com>] > > *Sent:* 05 September 2015 04:50 > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Subject:* Re: [Freeswitch-users] OneWay audio after hold->unhold > > > > We would be VERY appreciative if you could provide the configurations on > the Avaya and FreeSWITCH that allow them to talk to each other. In my > experience Avaya does not play well with others, and I have tried and given > up. > > Thank you in advance! > > Bote > > > > On Fri, Sep 4, 2015 at 12:09 PM, Prashanth Devarajappa < > Prashanth.Devarajappa(a)enghouse.com> wrote: > > Hello, > > > > I am connecting to Avaya switch using SIP trunk interface from FS with TLS > & SRTP connection. This works well for inbound and outbound calls. I have > an issue with media, when I try hold -> unhold operation from FS for an > outbound call. > > > > When FS tries to unhold the call by sending Re-INVITE, avaya responds with > 200 OK. Then sends INVITE w/o SDP to shuffle the media(IP Shuffling, to > make media flow directly b/w endpoints). FS responds with 200 OK + SDP as > expected. Now avaya sends ACK + SDP where SDP has the updated connection > info(IP and port of endpoint) where media should be sent. Along with this > it also changes the crypto master key as per RFC due to change in > connection info(c=). > > > > On seeing this ACK, FS update the connection info(which is good) and > change its own crypto master key in response. however this new master key > is not sent to far end, meaning FS can decode the media it receives but > other end can't decode the media FS is sending to it resulting in one way > audio. > > > > Has anyone come across this issue ? > > > > Is there any way to trigger Re-INVITE or UPDATE from FS so other end gets > new master key ? > > > > > > Regards > > Prashanth > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > ---------- Forwarded message ---------- > From: Stanislav Sinyagin <ssinyagin(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Sep 2015 08:55:37 +0200 > Subject: Re: [Freeswitch-users] Compiling under SmartOS > See the update at https://freeswitch.org/jira/browse/FS-7967 > > I fixed the compilation problems, and now there's a runtime issue. > > On Mon, Aug 17, 2015 at 10:40 AM, Stanislav Sinyagin > <ssinyagin(a)gmail.com> wrote: > > I see there are some people on the list, working with SmartOS. > > > > The current master fails to compile: > > https://freeswitch.org/jira/browse/FS-7967 > > > > Your input will be appreciated. > > > > I just started looking around and getting the feeling what SmartOS is. > > I worked with Solaris quite a lot, but that was almost 10 years ago. > > > > > ---------- Forwarded message ---------- > From: Prashanth Devarajappa <Prashanth.Devarajappa(a)enghouse.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Sep 2015 08:53:42 +0000 > Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold > > Hi Bob, > > > > I am told that our Avaya is “avaya communication manager is the 8300” > > > > So far with our testing, we have not seen any issues related to transfer. > (May be we have not done extensive transfer tests) > > > > Prashanth > > > > > > *From:* Bob Hartwig [mailto:bobjectsfreeswitch@gmail.com] > *Sent:* 07 September 2015 17:42 > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Subject:* Re: [Freeswitch-users] OneWay audio after hold->unhold > > > > Is this an Avaya AS5300 by any chance? Those have some serious interop > issues when it comes to transfers. I had to struggle mightily to get a > client's product (not Freeswitch) to interoperate with it. I'm really > interested to hear if this gets resolved for Freeswitch. > > > > > > > > On Fri, Sep 4, 2015 at 11:09 AM, Prashanth Devarajappa < > Prashanth.Devarajappa(a)enghouse.com> wrote: > > Hello, > > > > I am connecting to Avaya switch using SIP trunk interface from FS with TLS > & SRTP connection. This works well for inbound and outbound calls. I have > an issue with media, when I try hold -> unhold operation from FS for an > outbound call. > > > > When FS tries to unhold the call by sending Re-INVITE, avaya responds with > 200 OK. Then sends INVITE w/o SDP to shuffle the media(IP Shuffling, to > make media flow directly b/w endpoints). FS responds with 200 OK + SDP as > expected. Now avaya sends ACK + SDP where SDP has the updated connection > info(IP and port of endpoint) where media should be sent. Along with this > it also changes the crypto master key as per RFC due to change in > connection info(c=). > > > > On seeing this ACK, FS update the connection info(which is good) and > change its own crypto master key in response. however this new master key > is not sent to far end, meaning FS can decode the media it receives but > other end can't decode the media FS is sending to it resulting in one way > audio. > > > > Has anyone come across this issue ? > > > > Is there any way to trigger Re-INVITE or UPDATE from FS so other end gets > new master key ? > > > > > > Regards > > Prashanth > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)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(a)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(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 111, Issue 82
by kaleem rehman 08 Sep '15

08 Sep '15

Hi Prashanth / Bob, Avaya CM supports sip but this is mainly used for Aura VM, Experience portal and SM and more of Avaya only devices, direct conenctions to Asterisk , FS & 3CX work but I always experienced issues due to Avaya SIP Limitations and also officially this is not supported Best Route to do this is via session Manager as this is the supported method. First create a trunk link between Session manager and Freeswitch, then register extension directly to SM and make calls between from this extension and Freeswitch, once everything is working then create a trunk between CM & SM, or use existing and setup digit routing on SM and CM. if you have any issues setting up SM and its entities then let me know. this way will be best and far most easy one. thanks, kaleem On Tue, Sep 8, 2015 at 9:54 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: OneWay audio after hold->unhold (Bote Man) > 2. Re: Compiling under SmartOS (Stanislav Sinyagin) > 3. Re: OneWay audio after hold->unhold (Prashanth Devarajappa) > > > ---------- Forwarded message ---------- > From: Bote Man <bote_radio(a)botecomm.com> > To: "'FreeSWITCH Users Help'" <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 7 Sep 2015 18:03:17 -0400 > Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold > > Actually, yes I need all of the settings on the Avaya side. > > > > I am not using encryption and I never could get the Avaya CS1000M to talk > to the FreeSWITCH installation. I think the closest I could get was the > signaling would allow the call to be answered, but I got no audio to pass > either direction. > > > > I think the Avaya puts information in unexpected elements of the SIP > message and I had no more time to troubleshoot it because it was not my > primary task. > > > > There must be some small setting that I am missing and I would love to > find the missing piece of the puzzle. > > > > Thanks! > > > > Bote > > > > > > *From:* Prashanth Devarajappa > *Sent:* Monday, 07 September, 2015 09:47 > *Subject:* Re: [Freeswitch-users] OneWay audio after hold->unhold > > > > *Avaya PBX Version Control* > > § Media encryption over IP:Y > > § Session Manager Element Manager 6.3 > > § System Manager Release 6.3.5 > > § Communication System Management 6.3.10 > > > > *In IP-Network-region :* > > § Codec set matches group for encryption > > § Intra-region IP-IP Direct Audio: yes > > § Inter-region IP-IP Direct Audio: yes > > > > *In CM, signaling-group of type*: *SIP* > > § Transport Method: TLS > > > > Let me know if you need further Avaya config details. > > > > *FreeSwitch :* > > Its standard FS with custom (code/config) changes to enable > > > > · Support for RFC 5939 : Capability Negotiation > > · Support for UNENCRYPTED_SRTCP > > · Support IP Shuffling with in ACK…… to be done J > > > > > > Prashanth > > > > > > > > *From:* Bote Man [mailto:bote_radio@botecomm.com <bote_radio(a)botecomm.com>] > > *Sent:* 05 September 2015 04:50 > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Subject:* Re: [Freeswitch-users] OneWay audio after hold->unhold > > > > We would be VERY appreciative if you could provide the configurations on > the Avaya and FreeSWITCH that allow them to talk to each other. In my > experience Avaya does not play well with others, and I have tried and given > up. > > Thank you in advance! > > Bote > > > > On Fri, Sep 4, 2015 at 12:09 PM, Prashanth Devarajappa < > Prashanth.Devarajappa(a)enghouse.com> wrote: > > Hello, > > > > I am connecting to Avaya switch using SIP trunk interface from FS with TLS > & SRTP connection. This works well for inbound and outbound calls. I have > an issue with media, when I try hold -> unhold operation from FS for an > outbound call. > > > > When FS tries to unhold the call by sending Re-INVITE, avaya responds with > 200 OK. Then sends INVITE w/o SDP to shuffle the media(IP Shuffling, to > make media flow directly b/w endpoints). FS responds with 200 OK + SDP as > expected. Now avaya sends ACK + SDP where SDP has the updated connection > info(IP and port of endpoint) where media should be sent. Along with this > it also changes the crypto master key as per RFC due to change in > connection info(c=). > > > > On seeing this ACK, FS update the connection info(which is good) and > change its own crypto master key in response. however this new master key > is not sent to far end, meaning FS can decode the media it receives but > other end can't decode the media FS is sending to it resulting in one way > audio. > > > > Has anyone come across this issue ? > > > > Is there any way to trigger Re-INVITE or UPDATE from FS so other end gets > new master key ? > > > > > > Regards > > Prashanth > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > ---------- Forwarded message ---------- > From: Stanislav Sinyagin <ssinyagin(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Sep 2015 08:55:37 +0200 > Subject: Re: [Freeswitch-users] Compiling under SmartOS > See the update at https://freeswitch.org/jira/browse/FS-7967 > > I fixed the compilation problems, and now there's a runtime issue. > > On Mon, Aug 17, 2015 at 10:40 AM, Stanislav Sinyagin > <ssinyagin(a)gmail.com> wrote: > > I see there are some people on the list, working with SmartOS. > > > > The current master fails to compile: > > https://freeswitch.org/jira/browse/FS-7967 > > > > Your input will be appreciated. > > > > I just started looking around and getting the feeling what SmartOS is. > > I worked with Solaris quite a lot, but that was almost 10 years ago. > > > > > ---------- Forwarded message ---------- > From: Prashanth Devarajappa <Prashanth.Devarajappa(a)enghouse.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Sep 2015 08:53:42 +0000 > Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold > > Hi Bob, > > > > I am told that our Avaya is “avaya communication manager is the 8300” > > > > So far with our testing, we have not seen any issues related to transfer. > (May be we have not done extensive transfer tests) > > > > Prashanth > > > > > > *From:* Bob Hartwig [mailto:bobjectsfreeswitch@gmail.com] > *Sent:* 07 September 2015 17:42 > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Subject:* Re: [Freeswitch-users] OneWay audio after hold->unhold > > > > Is this an Avaya AS5300 by any chance? Those have some serious interop > issues when it comes to transfers. I had to struggle mightily to get a > client's product (not Freeswitch) to interoperate with it. I'm really > interested to hear if this gets resolved for Freeswitch. > > > > > > > > On Fri, Sep 4, 2015 at 11:09 AM, Prashanth Devarajappa < > Prashanth.Devarajappa(a)enghouse.com> wrote: > > Hello, > > > > I am connecting to Avaya switch using SIP trunk interface from FS with TLS > & SRTP connection. This works well for inbound and outbound calls. I have > an issue with media, when I try hold -> unhold operation from FS for an > outbound call. > > > > When FS tries to unhold the call by sending Re-INVITE, avaya responds with > 200 OK. Then sends INVITE w/o SDP to shuffle the media(IP Shuffling, to > make media flow directly b/w endpoints). FS responds with 200 OK + SDP as > expected. Now avaya sends ACK + SDP where SDP has the updated connection > info(IP and port of endpoint) where media should be sent. Along with this > it also changes the crypto master key as per RFC due to change in > connection info(c=). > > > > On seeing this ACK, FS update the connection info(which is good) and > change its own crypto master key in response. however this new master key > is not sent to far end, meaning FS can decode the media it receives but > other end can't decode the media FS is sending to it resulting in one way > audio. > > > > Has anyone come across this issue ? > > > > Is there any way to trigger Re-INVITE or UPDATE from FS so other end gets > new master key ? > > > > > > Regards > > Prashanth > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)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(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Re: [Freeswitch-users] OneWay audio after hold->unhold
by Prashanth Devarajappa 08 Sep '15

08 Sep '15

Thanks for your message. >> Best Route to do this is via session Manager as this is the supported method. First create a trunk link between Session manager and Freeswitch This is how we have setup our environment and got this far. Prashanth From: kaleem rehman [mailto:k4kaleem@gmail.com] Sent: 08 September 2015 10:46 To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 111, Issue 82 Hi Prashanth / Bob, Avaya CM supports sip but this is mainly used for Aura VM, Experience portal and SM and more of Avaya only devices, direct conenctions to Asterisk , FS & 3CX work but I always experienced issues due to Avaya SIP Limitations and also officially this is not supported Best Route to do this is via session Manager as this is the supported method. First create a trunk link between Session manager and Freeswitch, then register extension directly to SM and make calls between from this extension and Freeswitch, once everything is working then create a trunk between CM & SM, or use existing and setup digit routing on SM and CM. if you have any issues setting up SM and its entities then let me know. this way will be best and far most easy one. thanks, kaleem On Tue, Sep 8, 2015 at 9:54 AM, <freeswitch-users-request(a)lists.freeswitch.org<mailto:freeswitch-users-request@lists.freeswitch.org>> wrote: Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org<mailto:freeswitch-users-request@lists.freeswitch.org> You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org<mailto:freeswitch-users-owner@lists.freeswitch.org> When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: OneWay audio after hold->unhold (Bote Man) 2. Re: Compiling under SmartOS (Stanislav Sinyagin) 3. Re: OneWay audio after hold->unhold (Prashanth Devarajappa) ---------- Forwarded message ---------- From: Bote Man <bote_radio(a)botecomm.com<mailto:bote_radio@botecomm.com>> To: "'FreeSWITCH Users Help'" <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Cc: Date: Mon, 7 Sep 2015 18:03:17 -0400 Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold Actually, yes I need all of the settings on the Avaya side. I am not using encryption and I never could get the Avaya CS1000M to talk to the FreeSWITCH installation. I think the closest I could get was the signaling would allow the call to be answered, but I got no audio to pass either direction. I think the Avaya puts information in unexpected elements of the SIP message and I had no more time to troubleshoot it because it was not my primary task. There must be some small setting that I am missing and I would love to find the missing piece of the puzzle. Thanks! Bote From: Prashanth Devarajappa Sent: Monday, 07 September, 2015 09:47 Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold Avaya PBX Version Control • Media encryption over IP:Y • Session Manager Element Manager 6.3 • System Manager Release 6.3.5 • Communication System Management 6.3.10 In IP-Network-region : • Codec set matches group for encryption • Intra-region IP-IP Direct Audio: yes • Inter-region IP-IP Direct Audio: yes In CM, signaling-group of type: SIP • Transport Method: TLS Let me know if you need further Avaya config details. FreeSwitch : Its standard FS with custom (code/config) changes to enable • Support for RFC 5939 : Capability Negotiation • Support for UNENCRYPTED_SRTCP • Support IP Shuffling with in ACK…… to be done ☺ Prashanth From: Bote Man [mailto:bote_radio@botecomm.com] Sent: 05 September 2015 04:50 To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold We would be VERY appreciative if you could provide the configurations on the Avaya and FreeSWITCH that allow them to talk to each other. In my experience Avaya does not play well with others, and I have tried and given up. Thank you in advance! Bote On Fri, Sep 4, 2015 at 12:09 PM, Prashanth Devarajappa <Prashanth.Devarajappa(a)enghouse.com<mailto:Prashanth.Devarajappa@enghouse.com>> wrote: Hello, I am connecting to Avaya switch using SIP trunk interface from FS with TLS & SRTP connection. This works well for inbound and outbound calls. I have an issue with media, when I try hold -> unhold operation from FS for an outbound call. When FS tries to unhold the call by sending Re-INVITE, avaya responds with 200 OK. Then sends INVITE w/o SDP to shuffle the media(IP Shuffling, to make media flow directly b/w endpoints). FS responds with 200 OK + SDP as expected. Now avaya sends ACK + SDP where SDP has the updated connection info(IP and port of endpoint) where media should be sent. Along with this it also changes the crypto master key as per RFC due to change in connection info(c=). On seeing this ACK, FS update the connection info(which is good) and change its own crypto master key in response. however this new master key is not sent to far end, meaning FS can decode the media it receives but other end can't decode the media FS is sending to it resulting in one way audio. Has anyone come across this issue ? Is there any way to trigger Re-INVITE or UPDATE from FS so other end gets new master key ? Regards Prashanth _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto: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 ---------- Forwarded message ---------- From: Stanislav Sinyagin <ssinyagin(a)gmail.com<mailto:ssinyagin@gmail.com>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Cc: Date: Tue, 8 Sep 2015 08:55:37 +0200 Subject: Re: [Freeswitch-users] Compiling under SmartOS See the update at https://freeswitch.org/jira/browse/FS-7967 I fixed the compilation problems, and now there's a runtime issue. On Mon, Aug 17, 2015 at 10:40 AM, Stanislav Sinyagin <ssinyagin(a)gmail.com<mailto:ssinyagin@gmail.com>> wrote: > I see there are some people on the list, working with SmartOS. > > The current master fails to compile: > https://freeswitch.org/jira/browse/FS-7967 > > Your input will be appreciated. > > I just started looking around and getting the feeling what SmartOS is. > I worked with Solaris quite a lot, but that was almost 10 years ago. ---------- Forwarded message ---------- From: Prashanth Devarajappa <Prashanth.Devarajappa(a)enghouse.com<mailto:Prashanth.Devarajappa@enghouse.com>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Cc: Date: Tue, 8 Sep 2015 08:53:42 +0000 Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold Hi Bob, I am told that our Avaya is “avaya communication manager is the 8300” So far with our testing, we have not seen any issues related to transfer. (May be we have not done extensive transfer tests) Prashanth From: Bob Hartwig [mailto:bobjectsfreeswitch@gmail.com<mailto:bobjectsfreeswitch@gmail.com>] Sent: 07 September 2015 17:42 To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold Is this an Avaya AS5300 by any chance? Those have some serious interop issues when it comes to transfers. I had to struggle mightily to get a client's product (not Freeswitch) to interoperate with it. I'm really interested to hear if this gets resolved for Freeswitch. On Fri, Sep 4, 2015 at 11:09 AM, Prashanth Devarajappa <Prashanth.Devarajappa(a)enghouse.com<mailto:Prashanth.Devarajappa@enghouse.com>> wrote: Hello, I am connecting to Avaya switch using SIP trunk interface from FS with TLS & SRTP connection. This works well for inbound and outbound calls. I have an issue with media, when I try hold -> unhold operation from FS for an outbound call. When FS tries to unhold the call by sending Re-INVITE, avaya responds with 200 OK. Then sends INVITE w/o SDP to shuffle the media(IP Shuffling, to make media flow directly b/w endpoints). FS responds with 200 OK + SDP as expected. Now avaya sends ACK + SDP where SDP has the updated connection info(IP and port of endpoint) where media should be sent. Along with this it also changes the crypto master key as per RFC due to change in connection info(c=). On seeing this ACK, FS update the connection info(which is good) and change its own crypto master key in response. however this new master key is not sent to far end, meaning FS can decode the media it receives but other end can't decode the media FS is sending to it resulting in one way audio. Has anyone come across this issue ? Is there any way to trigger Re-INVITE or UPDATE from FS so other end gets new master key ? Regards Prashanth _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto: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(a)lists.freeswitch.org<mailto: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

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OneWay audio after hold->unhold
by Prashanth Devarajappa 08 Sep '15

08 Sep '15

Hello, I am connecting to Avaya switch using SIP trunk interface from FS with TLS & SRTP connection. This works well for inbound and outbound calls. I have an issue with media, when I try hold -> unhold operation from FS for an outbound call. When FS tries to unhold the call by sending Re-INVITE, avaya responds with 200 OK. Then sends INVITE w/o SDP to shuffle the media(IP Shuffling, to make media flow directly b/w endpoints). FS responds with 200 OK + SDP as expected. Now avaya sends ACK + SDP where SDP has the updated connection info(IP and port of endpoint) where media should be sent. Along with this it also changes the crypto master key as per RFC due to change in connection info(c=). On seeing this ACK, FS update the connection info(which is good) and change its own crypto master key in response. however this new master key is not sent to far end, meaning FS can decode the media it receives but other end can't decode the media FS is sending to it resulting in one way audio. Has anyone come across this issue ? Is there any way to trigger Re-INVITE or UPDATE from FS so other end gets new master key ? Regards Prashanth

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 111, Issue 80
by kaleem rehman 07 Sep '15

07 Sep '15

Hi Bob, what kit are you using on avaya side, is this a communication manager? Also what version u using. .. r u connecting directly to CM or via session manager. If CM, have u checked IP network region where you placed freeswitch and if direct ip to ip on inter region is disabled Send the teunk config and I will be able to help u on communication manager or session manager side. Thanks Kaleem On 7 Sep 2015 18:37, <freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: OneWay audio after hold->unhold (Bob Hartwig) > 2. Re: FreeSwitch - Performance issues (Michael Giagnocavo) > > > ---------- Forwarded message ---------- > From: Bob Hartwig <bobjectsfreeswitch(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 7 Sep 2015 11:41:55 -0500 > Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold > Is this an Avaya AS5300 by any chance? Those have some serious interop > issues when it comes to transfers. I had to struggle mightily to get a > client's product (not Freeswitch) to interoperate with it. I'm really > interested to hear if this gets resolved for Freeswitch. > > > > On Fri, Sep 4, 2015 at 11:09 AM, Prashanth Devarajappa < > Prashanth.Devarajappa(a)enghouse.com> wrote: > >> Hello, >> >> >> >> I am connecting to Avaya switch using SIP trunk interface from FS with >> TLS & SRTP connection. This works well for inbound and outbound calls. I >> have an issue with media, when I try hold -> unhold operation from FS for >> an outbound call. >> >> >> >> When FS tries to unhold the call by sending Re-INVITE, avaya responds >> with 200 OK. Then sends INVITE w/o SDP to shuffle the media(IP Shuffling, >> to make media flow directly b/w endpoints). FS responds with 200 OK + SDP >> as expected. Now avaya sends ACK + SDP where SDP has the updated connection >> info(IP and port of endpoint) where media should be sent. Along with this >> it also changes the crypto master key as per RFC due to change in >> connection info(c=). >> >> >> >> On seeing this ACK, FS update the connection info(which is good) and >> change its own crypto master key in response. however this new master key >> is not sent to far end, meaning FS can decode the media it receives but >> other end can't decode the media FS is sending to it resulting in one way >> audio. >> >> >> >> Has anyone come across this issue ? >> >> >> >> Is there any way to trigger Re-INVITE or UPDATE from FS so other end gets >> new master key ? >> >> >> >> >> >> Regards >> >> Prashanth >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > ---------- Forwarded message ---------- > From: Michael Giagnocavo <mgg(a)giagnocavo.net> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 7 Sep 2015 17:35:49 +0000 > Subject: Re: [Freeswitch-users] FreeSwitch - Performance issues > > Email me direct (mgg(a)giagnocavo.net) next week and I’ll help out on this > if you’d like. > > > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Bote Man > *Sent:* Friday, September 4, 2015 7:53 AM > *To:* 'FreeSWITCH Users Help' <freeswitch-users(a)lists.freeswitch.org> > *Subject:* Re: [Freeswitch-users] FreeSwitch - Performance issues > > > > This thread contains a great deal of useful intelligence that would be a > shame to lose. > > > > There is a recent discussion of developing a proven High Availability > configuration for FreeSWITCH; might there be a better way to track these > findings so that they can be documented as part of that effort, both by the > Documentation Team and by those doing the research? This is really good > stuff. > > > > My personal time is being consumed close to 100% right now so I won’t be > able to do anything with it until next week. > > > > Thanks. > > > > Bote > > Docs Team guy > > > > > > *From:* Shaun Stokes > *Sent:* Friday, 04 September, 2015 09:25 > *Subject:* Re: [Freeswitch-users] FreeSwitch - Performance issues > > > > Hi Michael, > > > > We’re now seeing suitable performance on Hyper-V, but as we’re processing > media\RTP we will be moving to physical hardware. If we need FreeSwitch > servers to process SIP messages only we will consider these for > virtualization. > > > > Thanks for your help, > > Shaun > > > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [ > mailto:freeswitch-users-bounces@lists.freeswitch.org > <freeswitch-users-bounces(a)lists.freeswitch.org>] *On Behalf Of *Michael > Giagnocavo > *Sent:* 02 September 2015 21:19 > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Subject:* Re: [Freeswitch-users] FreeSwitch - Performance issues > > > > Years back, I used to run FS **on Windows**, under Hyper-V (2008), on a > Q6600 and had considerably higher perf numbers. Hundreds of CPS (shit > traffic), with 1000-2000 in progress channels (no media, ASR like 10% > maybe). So something is certainly pathologically wrong if you’re seeing > things get maxed out at the numbers you show. I didn’t do anything special, > and did routing both in-process via mod_managed as well as xml_curl. > > > > Do you have a reason not to put the FS core SQLite db on a tmpfs? That’s a > big win. Also make sure FS’s temp dir is a tmpfs. Unless you’re persisting > and using stuff between sessions, there’s no point in going to disk right? > And your random, small, IOPS are gonna be an order of magnitude less than > what you posted, at least. > > > > One thing to be aware of is CPU pinning and scheduling. First, on > multi-socket systems, you have NUMA. This really hurts. Raw memory > bandwidth takes a 40% hit. More importantly, you’re toasting your caches > (which with tons of threads happens a lot anyways). One number I like to > reference is that single ownership transfer of one cache line from socket > to another takes ~300 cycles (vs ~10 cycles for L2 and ~40 cycles for L3). > Hyper-V has an option to lock servers to a single NUMA node. Make sure this > is enabled (it may cause VMs not to start if it cannot allocate all the > resources requested – this is a good thing for VoIP apps). Then cap the > individual VMs so they aren’t scheduling all over the place (34 should not > need more than 1 CPU). FS spawns tons of threads and they’ll end up running > on all CPUs even if they don’t need all that time, and that then screws up > cache usage even more. But more importantly, it sounds like this is a busy > server? Are you sure you’re not just way oversubscribed and the VM is > losing out? Your steal% is 0 so I’m guessing HV isn’t reporting that > accurately. > > > > Only other thing that comes to mind is that Hyper-V does terrible with > time, so if you’re relying on HV time service you’re gonna be skewing all > over. I don’t think this should hurt FS directly, but you should run ntp > anyways to be safe. > > > > If you end up needing to scale up on a single server, you’re better off > running multiple FS instances, each pinned to specific cores. And you > should have OpenSIPS or something in front to do LB (and it provides a ton > of flexibility and handiness anyways). This probably only matters if you’re > doing high CPS (dialer or shit traffic that mostly isn’t completing). I’m > not sure what exactly you’re benching, but it sounds like your actual use > case is different than the test, eh? > > > > Email me directly and we can chat more about this. > > > > -Michael > > > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [ > mailto:freeswitch-users-bounces@lists.freeswitch.org > <freeswitch-users-bounces(a)lists.freeswitch.org>] *On Behalf Of *Shaun > Stokes > *Sent:* Wednesday, September 2, 2015 5:09 AM > *To:* 'FreeSWITCH Users Help' <freeswitch-users(a)lists.freeswitch.org> > *Subject:* Re: [Freeswitch-users] FreeSwitch - Performance issues > > > > Hi Jay, > > > > Thanks for the suggestion. > > > > We’re actually using PostgreSQL, the file system is a 2TB RAID5 with 10x > 278GB SAS 10K RPM disks, read is 1GB/s (23,000 IOPS) write is 500MB/s > (15,000 IOPS) but we’re only using a fraction of this. > > > > Thanks, > > Shaun > > > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [ > mailto:freeswitch-users-bounces@lists.freeswitch.org > <freeswitch-users-bounces(a)lists.freeswitch.org>] *On Behalf Of *jay binks > *Sent:* 02 September 2015 11:37 > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Subject:* Re: [Freeswitch-users] FreeSwitch - Performance issues > > > > I may have missed it in your previous emails, but im assuming you have the > core DB on a tempfs > or somewhere other than the default location on spinning rust ? > > > > Jay > > > > On 2 September 2015 at 18:53, Shaun Stokes < > shaun.stokes(a)itec-support.co.uk> wrote: > > Hi Brian, > > > > I’m a huge fan of virtualization, but in our environment it simply doesn’t > provide the performance we’re looking for. The tests carried out were 10 > second calls which were routed to a Call Centre Queue with one registered > SIP extension as a member. > > > > There was a significant improvement when we tested on an Amazon VM in > comparison to Hyper-V, but we’d like a system which can handle large > volumes of call traffic, based on our tests this will need to be physical > hardware to achieve the best performance. > > > > Further to this, we have managed to resolve our initial problem with CPU > spikes on Hyper-V following these steps: > > · Disable VMQ on all physical and virtual adapters on the Hyper-V > host which are linked to the FreeSwitch VM > > · Reduce the virtual cores on all VMs to at least half of the > available physical cores > > · Increase the Processor Relative weight on the FreeSwitch VMs to > 200 (default 100) > > > > Thanks, > > Shaun > > > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* 01 September 2015 21:21 > > > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Subject:* Re: [Freeswitch-users] FreeSwitch - Performance issues > > > > Shaun, > > > > 800 concurrent with 400cps, expectations are a little out of touch with > reality and your test is no where near real world. Your call duration is > ~2000ms, little to no time to actually do anything useful. You're burning > the bulk of your CPU time on signaling and doing no real work with the > session. > > > > So yes, you'll always get the best results out of bare metal, but you can > also get good performance on a VM (*) > > > > (*) This is in no way an endorsement of running FreeSWITCH in a VM, Your > mileage may vary depending on your specific application and load > requirements. > > > > Thanks, > > > > > > > > > > On Tue, Sep 1, 2015 at 3:08 PM, Shaun Stokes < > shaun.stokes(a)itec-support.co.uk> wrote: > > Update > > We've performed various tests and have come to the conclusion as advised > by others, that visualization is causing bottlenecks. > > Using Amazon EC2 on an 8 core high performance virtual server this maxed > out at around 800 concurrent sessions, 400 sessions per second. > > Using a Dell 1950 8 core server this was able to handle 1000 concurrent > sessions, 400 sessions per second and still have plenty of resource > available. Average CPU was fluctuating between 30-40% under peak load. > > Conclusion, if you want to run a high performance FreeSwitch server in a > production you should stick with physical hardware and avoid visualization > where possible. > ________________________________________ > From: freeswitch-users-bounces(a)lists.freeswitch.org [ > freeswitch-users-bounces(a)lists.freeswitch.org] on behalf of Shaun Stokes [ > shaun.stokes(a)itec-support.co.uk] > Sent: 28 August 2015 15:37 > To: 'FreeSWITCH Users Help' > > Subject: Re: [Freeswitch-users] FreeSwitch - Performance issues > > Hi Nathan, > > Thanks for the suggestion, we still see the CPU spikes when is under load > after reducing to 8 cores but I think there may actually be an improvement > in the audio quality. > > It's still not what we're expecting, so we will have to test our build on > physical hardware too. > > Thanks, > Shaun > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] On Behalf Of Nathan > Neulinger > Sent: 28 August 2015 15:04 > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Subject: Re: [Freeswitch-users] FreeSwitch - Performance issues > > Yes, I know you said Hyper-V, I don't know if it has the same sort of > scheduler behavior or not, just mentioned the vmware for context. > > On 08/28/2015 08:46 AM, Nathan Neulinger wrote: > > One thing that I see jump out - you've probably got too many virtual > > cores defined on that VM, particularly since you've assigned # vcpu the > same as the number of real cores on the host system. > > > > Try your test again dropping that to 8 virtual CPUs and I think you > > might see better results. (Or at least drop to less than 100% of the > > total cores on the hardware.) > > > > Particularly if this host system has other vms running on it, you're > > hurting the vm scheduler's ability to hand out resources. With vmware, > > I believe a simple rule of thumb is that scheduler requires N-1 CPUs > > to be locked to schedule execution of an N cpu virtual machine - so even > if it doesn't need all of the CPUs, it has to reserve all of them any time > you have more than a 2 vCPU vm. > > > > -- Nathan > > > > On 08/28/2015 08:18 AM, Michael Jerris wrote: > >> The numbers your posting are fairly low, I'm guessing this is a > >> grossly underpowered vm that is causing your issues > >> > >> On Friday, August 28, 2015, Shaun Stokes < > shaun.stokes(a)itec-support.co.uk <mailto:shaun.stokes@itec-support.co.uk>> > wrote: > >> > >> Hi Peter,____ > >> > >> __ __ > >> > >> Thanks for the advice, this is something we're already looking > into but we don't have the new hardware available > >> yet.____ > >> > >> __ __ > >> > >> Debian 8 has great integration with Hyper-V on 2012R2 and operates > as a Generation 2 VM, obviously it's never going > >> to be as good as running directly on the hardware but we were > >> hoping for better performance.____ > >> > >> __ __ > >> > >> Thanks,____ > >> > >> Shaun____ > >> > >> __ __ > >> > >> *From:*freeswitch-users-bounces@lists.freeswitch.org > >> <javascript:_e(%7B%7D,'cvml',' > freeswitch-users-bounces(a)lists.freeswitch.org');> > >> [mailto:freeswitch-users-bounces@lists.freeswitch.org > >> <javascript:_e(%7B%7D,'cvml',' > freeswitch-users-bounces(a)lists.freeswitch.org');>] *On Behalf Of *Peter > Olsson > >> *Sent:* 28 August 2015 10:43 > >> *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org > >> <javascript:_e(%7B%7D,'cvml',' > freeswitch-users(a)lists.freeswitch.org');>> > >> *Subject:* Re: [Freeswitch-users] FreeSwitch - Performance > >> issues____ > >> > >> __ __ > >> > >> If you want high performance, I recommend using real hardware, not > a virtual machine setup - it will cause you > >> issues.____ > >> > >> __ __ > >> > >> /Peter____ > >> > >> __ __ > >> > >> 2015-08-28 9:44 GMT+02:00 Shaun Stokes < > shaun.stokes(a)itec-support.co.uk > >> > >> <javascript:_e(%7B%7D,'cvml','shaun.stokes(a)itec-support.co.uk');>>:__ > >> __ > >> > >> Hi Michael,____ > >> > >> ____ > >> > >> Thanks for the response and recommendation.____ > >> > >> ____ > >> > >> Our new build is now on Debian 8 Jessie, performance on the > system is noticeably better and any audio problems > >> while the system is under load are significantly reduced but > we do still experience spikes along with reduced > >> audio quality when the system is processing 20 calls per > >> second while already maintaining 80 existing calls.____ > >> > >> ____ > >> > >> We're still hitting some kind of bottleneck, what should we > expect to be able to support on a single piece of > >> hardware? We've kept our dialplans as small as possible and > we're using Memcache, but something tells me it's > >> the call setup process which is triggering the spikes, is this > possibly a limitation on the SIP profiles which > >> are single threaded?____ > >> > >> ____ > >> > >> I also see FreeSWITCH supports clustering but it's in relation > to high availability, is it possible to cluster a > >> single FreeSWITCH instance across multiple hardware or should > we be looking elsewhere to solve these > >> bottlenecks?____ > >> > >> ____ > >> > >> We switched to TCP Vegas on our new build as recommended on > the FreeSWITCH Wiki, this has provided a noticeable > >> reduction in audio latency but doesn't solve the issue with > >> CPU spikes.____ > >> > >> ____ > >> > >> Many Thanks,____ > >> > >> Shaun____ > >> > >> ____ > >> > >> *From:*freeswitch-users-bounces@lists.freeswitch.org > >> <javascript:_e(%7B%7D,'cvml',' > freeswitch-users-bounces(a)lists.freeswitch.org');> > >> [mailto:freeswitch-users-bounces@lists.freeswitch.org > >> <javascript:_e(%7B%7D,'cvml',' > freeswitch-users-bounces(a)lists.freeswitch.org');>] *On Behalf Of *Michael > Jerris > >> *Sent:* 15 August 2015 23:00 > >> *To:* FreeSWITCH Users Help < > freeswitch-users(a)lists.freeswitch.org > >> <javascript:_e(%7B%7D,'cvml',' > freeswitch-users(a)lists.freeswitch.org');>> > >> *Subject:* Re: [Freeswitch-users] FreeSwitch - Performance > >> issues____ > >> > >> ____ > >> > >> There are multiple known issues with Ubuntu 12.04. We > reccomend Debian 8 Jessie. > >> > >> On Friday, August 14, 2015, Shaun Stokes < > shaun.stokes(a)itec-support.co.uk > >> > >> <javascript:_e(%7B%7D,'cvml','shaun.stokes(a)itec-support.co.uk');>> > >> wrote:____ > >> > >> Hi,____ > >> > >> ____ > >> > >> We're experiencing performance issues with FreeSwitch, our > target is 500 concurrent sessions, but at the > >> moment this starts to bottleneck around 30.____ > >> > >> ____ > >> > >> Host system:____ > >> > >> Windows Server 2012 Hyper-V Host____ > >> > >> AMD Opteron 4386 (2 processors) - 16 cores total____ > >> > >> 128GB DDR3____ > >> > >> 2TB RAID 5 (700MB/s tested read and write > >> throughput)____ > >> > >> ____ > >> > >> FreeSwitch Virtual Machine:____ > >> > >> FreeSwitch 1.4.15____ > >> > >> Ubuntu 12.04 LTS____ > >> > >> 16 Virtual cores (high priority)____ > >> > >> 2GB RAM (would assign more but FreeSwitch never seems to > >> use much)____ > >> > >> 500GB HD (on VHDX)____ > >> > >> ____ > >> > >> After around 30 concurrent sessions we begin to see CPU > spikes almost every time a new call comes in, as the > >> sessions increase the size and frequency of the CPU spikes > also increase. The system seems to be able to sit > >> comfortably with over 100 concurrent sessions and 80% idle > CPU, providing we don't have any new calls > >> hitting the platform. The spikes are causing audio (RTP) > to stutter or in some cases drop completely for a > >> few seconds.____ > >> > >> ____ > >> > >> The FreeSwitch spikes are occurring on all 16 cores, we > have been monitoring the system using htop and > >> mpstat.____ > >> > >> ____ > >> > >> This is an example of when we receive an inbound call > >> while we have 34 concurrent sessions:____ > >> > >> 14:26:04 CPU %usr %nice %sys %iowait %irq > %soft %steal %guest %idle____ > >> > >> 14:26:05 all 1.67 0.00 27.34 0.00 0.25 > 0.00 0.00 0.00 70.74____ > >> > >> 14:26:05 0 1.00 0.00 28.00 0.00 0.00 > 0.00 0.00 0.00 71.00____ > >> > >> 14:26:05 1 0.99 0.00 26.73 0.00 0.00 > 0.00 0.00 0.00 72.28____ > >> > >> 14:26:05 2 0.99 0.00 26.73 0.00 0.00 > 0.00 0.00 0.00 72.28____ > >> > >> 14:26:05 3 3.92 0.00 26.47 0.00 0.00 > 0.00 0.00 0.00 69.61____ > >> > >> 14:26:05 4 0.99 0.00 25.74 0.00 1.98 > 0.00 0.00 0.00 71.29____ > >> > >> 14:26:05 5 0.99 0.00 26.73 0.00 0.99 > 0.00 0.00 0.00 71.29____ > >> > >> 14:26:05 6 0.00 0.00 26.47 0.00 0.98 > 0.00 0.00 0.00 72.55____ > >> > >> 14:26:05 7 0.00 0.00 26.73 0.00 0.00 > 0.00 0.00 0.00 73.27____ > >> > >> 14:26:05 8 3.96 0.00 27.72 0.00 0.00 > 0.00 0.00 0.00 68.32____ > >> > >> 14:26:05 9 2.00 0.00 27.00 0.00 0.00 > 0.00 0.00 0.00 71.00____ > >> > >> 14:26:05 10 6.00 0.00 32.00 0.00 0.00 > 0.00 0.00 0.00 62.00____ > >> > >> 14:26:05 11 1.96 0.00 27.45 0.00 0.00 > 0.00 0.00 0.00 70.59____ > >> > >> 14:26:05 12 0.99 0.00 26.73 0.00 0.00 > 0.00 0.00 0.00 72.28____ > >> > >> 14:26:05 13 1.00 0.00 28.00 0.00 0.00 > 0.00 0.00 0.00 71.00____ > >> > >> 14:26:05 14 0.00 0.00 27.00 0.00 0.00 > 0.00 0.00 0.00 73.00____ > >> > >> ____ > >> > >> This is when the system is not receiving an inbound call > but is sitting comfortably at 34 concurrent > >> sessions:____ > >> > >> 14:25:57 CPU %usr %nice %sys %iowait %irq > %soft %steal %guest %idle____ > >> > >> 14:25:58 all 0.87 0.00 0.62 0.00 0.12 > 0.00 0.00 0.00 98.39____ > >> > >> 14:25:58 0 0.99 0.00 0.99 0.00 0.00 > 0.00 0.00 0.00 98.02____ > >> > >> 14:25:58 1 0.99 0.00 0.99 0.00 0.00 > 0.00 0.00 0.00 98.02____ > >> > >> 14:25:58 2 0.98 0.00 0.98 0.00 0.00 > 0.00 0.00 0.00 98.04____ > >> > >> 14:25:58 3 0.00 0.00 0.99 0.00 0.00 > 0.00 0.00 0.00 99.01____ > >> > >> 14:25:58 4 0.00 0.00 0.00 0.00 1.98 > 0.00 0.00 0.00 98.02____ > >> > >> 14:25:58 5 0.00 0.00 0.00 0.00 0.99 > 0.00 0.00 0.00 99.01____ > >> > >> 14:25:58 6 0.00 0.00 0.99 0.00 0.00 > 0.00 0.00 0.00 99.01____ > >> > >> 14:25:58 7 0.00 0.00 0.99 0.00 0.00 > 0.00 0.00 0.00 99.01____ > >> > >> 14:25:58 8 5.00 0.00 1.00 0.00 0.00 > 0.00 0.00 0.00 94.00____ > >> > >> 14:25:58 9 1.00 0.00 0.00 0.00 0.00 > 0.00 0.00 0.00 99.00____ > >> > >> 14:25:58 10 1.00 0.00 0.00 0.00 0.00 > 0.00 0.00 0.00 99.00____ > >> > >> 14:25:58 11 0.99 0.00 0.99 0.00 0.00 > 0.00 0.00 0.00 98.02____ > >> > >> 14:25:58 12 0.00 0.00 0.00 0.00 0.00 > 0.00 0.00 0.00 100.00____ > >> > >> 14:25:58 13 1.98 0.00 0.99 0.00 0.00 > 0.00 0.00 0.00 97.03____ > >> > >> 14:25:58 14 0.00 0.00 0.00 0.00 0.00 > 0.00 0.00 0.00 100.00____ > >> > >> 14:25:58 15 0.00 0.00 0.00 0.00 0.00 > 0.00 0.00 0.00 100.00____ > >> > >> ____ > >> > >> We're using FreeSwitch in multi-tenant mode, we have tried > dedicating a single VM for one tenancy but still > >> experience the issue. My assumption has been that this > will be due to the dialplan, I've optimized this > >> slightly by writing LUA scripts to handle some of the > inbound calls this appears to take away some of the > >> load but we're still using the internal dialplans for each > tenancy (tenancy dialplans have an average of 300 > >> entries).____ > >> > >> ____ > >> > >> We are using the following arguments when running > >> FreeSwitch:____ > >> > >> -rp -nc -nonat____ > >> > >> ____ > >> > >> I've seen another post which says we should operate > FreeSwitch in High Performance mode using -hp but can't > >> find anything information about this, is this still a > >> valid argument to use with FreeSwitch?____ > >> > >> ____ > >> > >> Has anyone experienced similar performance issues before > >> or have any advice?____ > >> > >> ____ > >> > >> Hope someone may be able help.____ > >> > >> ____ > >> > >> Thanks,____ > >> > >> Shaun____ > >> > >> > >> > ______________________________________________________________________ > >> This message has been checked for all known viruses by > MessageLabs Virus Scanning Service. > >> > >> _____________________________________________________________________ > >> _____ > >> > >> > >> > ______________________________________________________________________ > >> This message has been checked for all known viruses by > MessageLabs Virus Scanning Service. > >> > >> _____________________________________________________________________ > >> _____ > >> > >> > >> > ______________________________________________________________________ > >> This message has been checked for all known viruses by > MessageLabs Virus Scanning Service. > >> > >> _____________________________________________________________________ > >> _____ > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting(a)freeswitch.org <javascript:_e(%7B%7D,'cvml',' > consulting(a)freeswitch.org');> > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users(a)lists.freeswitch.org > <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users(a)lists.freeswitch.org');> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org____ > >> > >> __ __ > >> > >> > >> > ______________________________________________________________________ > >> This message has been checked for all known viruses by MessageLabs > Virus Scanning Service. > >> > >> _____________________________________________________________________ > >> _____ > >> > >> > >> > ______________________________________________________________________ > >> This message has been checked for all known viruses by MessageLabs > Virus Scanning Service. > >> > >> _____________________________________________________________________ > >> _ > >> > >> > >> > >> _____________________________________________________________________ > >> ____ Professional FreeSWITCH Consulting Services: > >> consulting(a)freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users(a)lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us > >> ers > >> http://www.freeswitch.org > >> > > > > -- > ------------------------------------------------------------ > Nathan Neulinger nneul(a)mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > > *Brian West* > brian(a)freeswitch.org > > [image: http://billing.freeswitch.org/templates/default/img/whmcslogo.png] > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: > /r/freeswitch <https://www.reddit.com/r/freeswitch> > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > > Sincerely > > Jay > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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