[Freeswitch-users] How to create a single outbound ESL socket at freeswitch startup.

Chris Cachor ccachor at gmail.com
Mon Aug 12 19:06:26 MSD 2013 

Julien,

Unfortunately I've run into the same problem. Having an outbound ESL connection to push event data to a ESL server is ideal, especially in a multi-server environment (simple config setup). However, if you read the documentation, the intended behavior of the outbound ESL socket is to give control of the call to the ESL server/application. If that's your implementation, then you could simply add the socket application to your dial plan. Otherwise, an inbound socket connection is what you're looking for. 

- Chris

On Aug 12, 2013, at 9:46 AM, freeswitch-users-request at lists.freeswitch.org wrote:

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>   1. How to create a single outbound ESL socket at	freeswitch
>      startup. (julien terrasson)
>   2.  uuid_displace: change loudness level of audio
>      (Melanie Treitinger)
>   3. Conference for Flash and WebRTC users (pablo platt)
>   4. Re: Error in launching fs_cli (Ashish Mishra)
>   5. Re: garbled audio with G726-32,	other codecs are fine
>      (Michael Jerris)
>   6. Re: Error in launching fs_cli (Steven Ayre)
>   7. Re: Issue with JSSIP + Freeswitch (Michael Jerris)
> 
> From: julien terrasson <julien.terrasson at gmail.com>
> Subject: [Freeswitch-users] How to create a single outbound ESL socket at freeswitch startup.
> Date: August 12, 2013 2:01:37 AM CDT
> To: freeswitch-users at lists.freeswitch.org
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> Hi,
> 
> I would like to create a single outbound ESL socket that would establish the connection with the ESL server at freeswitch startup.
> This connexion would remains up and would used to transfer events from calls going through this server.
> I managed to do this with an inbound ESL but i would prefer doing it with an outbound ESL (it is important for me that freeswitch controls the establishement of the connection).
> Does anybody know if that's possible to start a common ESL connection (that would push event from all call) at FS startup ? If yes, where should it be provisionned (from the dialplan? in a startup script ?)
> 
> Hope somebody can help..
> 
> Regards,
> 
> Julien
> 
> 
> 
> 
> 
> 
> 
> 
> From: Melanie Treitinger <treitinger at as-infodienste.de>
> Subject: [Freeswitch-users] uuid_displace: change loudness level of audio
> Date: August 12, 2013 3:33:36 AM CDT
> To: freeswitch-users at lists.freeswitch.org
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> Hi Michael,
> 
> I think I'll try sox, thanks.
> 
> 
> Melanie
> 
> 
> 
> From: pablo platt <pablo.platt at gmail.com>
> Subject: [Freeswitch-users] Conference for Flash and WebRTC users
> Date: August 12, 2013 2:40:17 AM CDT
> To: freeswitch-users at lists.freeswitch.org
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> Hi,
> 
> I'm currently using a RTMP server for audio conference.
> There are no performance or latency issues because there is no encryption, transcoding or mixing involved. A 1GB VPS server can handle more than 100 users without a problem.
> 
> I'm trying to evaluate FreeSWITCH so I'll be able to support both Flash and WebRTC users in a conference.
> 
> My use case is two types of conferences:
> - A conference with up to 5 participants all speaking.
> - A conference with 20-30 participants with 2-3 speakers.
> 
> What part require more CPU?
> - Encryption (DTLS-SRTP)
> - Transcoding
> - Mixing
> 
> Does FS mix a channel separately for each participant or can it reuse the same mix?
> For example, if I have 1 speaker and 20 listeners, 10 with speex and 10 with opus, does FS produce 2 streams or 20?
> 
> Does FS exclude silent participants from the mix to save CPU?
> 
> Thanks
> 
> 
> 
> 
> 
> 
> From: Ashish Mishra <itsme.kunnu at gmail.com>
> Subject: Re: [Freeswitch-users] Error in launching fs_cli
> Date: August 12, 2013 3:51:45 AM CDT
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> I reinstalled freeswitch but still i am getting the same error :
> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ] mod_socket is also loaded...i have also flushed the firewall rules...pls help
> 
> On Aug 6, 2013 2:30 AM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which i have installed freeswitch) it gives me the following error :
> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ] 
> 
> 
> 
> From: Michael Jerris <mike at jerris.com>
> Subject: Re: [Freeswitch-users] garbled audio with G726-32, other codecs are fine
> Date: August 12, 2013 9:14:45 AM CDT
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> There is always bit packing, but there are 2 different ways to do the bit packing.  A lot of devices get it wrong so its worth looking at that.
> 
> Mike
> 
> On Aug 4, 2013, at 1:53 AM, Ivan Mitev <imitev at c3i.bg> wrote:
> 
>> Thanks for the suggestion but I'm testing with G726-32, not AAL2-G726-32 
>> ; so bitpacking shouldn't be used. By the way when I tested with AAL2 to 
>> the linphone client I only got cracks and whitenoise, I've forgot to 
>> mention that in my post.
>> 
>> That said I've tried to uncomment and set <param name="bitpacking" 
>> value="none"/> in internal.xml ("none" is a wild guess - I couldn't find 
>> any doc on values accepted by this parameter), but that doesn't help.
>> 
>> Speex: the ATAs don't support it. And being stubborn I'd like to 
>> understand what's wrong with G726 :)
>> 
>> 
>> On 08/04/2013 05:22 AM, Jeff Leung wrote:
>>> 
>>> You can turn off G726 AAC bit-packing in spandsp.conf.xml.
>>> 
>>> By the way, there are other codecs out there you can try. SPEEX comes 
>>> to mind if all your endpoints don’t deal with the PSTN.
>>> 
>>> *From:*freeswitch-users-bounces at lists.freeswitch.org 
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
>>> *Brian Foster
>>> *Sent:* Saturday, August 3, 2013 2:37 PM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] garbled audio with G726-32, other 
>>> codecs are fine
>>> 
>>> AAC bitpacking by any chance? I thought I had a similar issue, 
>>> happened so long ago I cant remember what I did.
>>> 
>>> Thank you,
>>> 
>>> Brian Foster
>>> Project Manager/Owner's Rep.
>>> Davri Investments, Inc.
>>> O: 317-787-2686 x2102
>>> M: 317-600-9753
>>> E: bdfoster at davri.com <mailto:bdfoster at davri.com>
>>> Indianapolis, Indiana
>>> 
>>> Sent from a mobile device.
>>> 
>>> On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev at c3i.bg 
>>> <mailto:imitev at c3i.bg>> wrote:
>>> 
>>> Hello
>>> 
>>> I'm migrating an office setup from asterisk to FS and in the process I
>>> was considering using G726-32 for some bandwidth starved remote
>>> endpoints. However I only get metallic/garbled audio with that codec
>>> even when simply playing moh to the endpoint, while other codecs work
>>> fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds
>>> marginally better but still garbled and really worse than G711.
>>> 
>>> The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest
>>> (centos6 64bit host). But please don't shoot ! :) - I know about virtual
>>> environment limitations but for these tests the host is only lightly
>>> loaded, there aren't any calls to the FS instance except my tests, and
>>> the fact that it works with other codecs makes me think that
>>> virtualization is not the issue here. I may be wrong though.
>>> 
>>> Is there any guide for debugging that kind of problem before reverting
>>> to a fresh install on bare-metal with the latest HEAD ? Until now I've
>>> tried:
>>> 
>>> - improving timers ; but the default soft timer (which I guess uses
>>> timerd) works best. The time interval between sent packets on a tcpdump
>>> trace looks identical to the output of "timer_test", so that doesn't
>>> seem to be a network/jitter problem. And there's no problem with other
>>> codecs, but maybe G726-XX is specific. For info the guest's clocksource
>>> is kvm_clock, while the host uses tsc.
>>> 
>>> - using different endpoints: the production ones are Linksys PAP2
>>> ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the
>>> same thing happens with linphone on a fedora 19 laptop.
>>> 
>>> A call with rtp media going through FS without transcoding - G726-32 to
>>> G726-32 - works perfectly (I can't hear the difference with G711). The
>>> problem is only when there's transcoding to G726 (from wav for moh, or
>>> from any other codec when bridging). I've looked at the wiki, posts,
>>> changelogs, jira, ..., but am a bit at a loss now.
>>> 
>>> Any pointers ?
>>> 
>>> Except that little problem, FS rocks, and I'm happy I can finally ditch
>>> asterisk. Kudos to the core devs and contributors.
>>> 
>>> Ivan
>>> 
> 
> 
> 
> 
> 
> 
> From: Steven Ayre <steveayre at gmail.com>
> Subject: Re: [Freeswitch-users] Error in launching fs_cli
> Date: August 12, 2013 9:44:38 AM CDT
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> Have you checked the logfile and netstat?
> 
> 
> On 12 August 2013 09:51, Ashish Mishra <itsme.kunnu at gmail.com> wrote:
> I reinstalled freeswitch but still i am getting the same error :
> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ] mod_socket is also loaded...i have also flushed the firewall rules...pls help
> 
> On Aug 6, 2013 2:30 AM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which i have installed freeswitch) it gives me the following error :
> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ] 
> 
> _________________________________________________________________________
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> 
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> 
> 
> From: Michael Jerris <mike at jerris.com>
> Subject: Re: [Freeswitch-users] Issue with JSSIP + Freeswitch
> Date: August 12, 2013 9:45:46 AM CDT
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> We only use non rfc-1918 ip's by default.  If you want to use 1918 ip's you need to tweak acls
> 
> Mike
> 
> 
> On Aug 5, 2013, at 8:24 AM, Shahrzad A. <shahrzad.aziminia at gmail.com> wrote:
> 
>> Hi  everyone
>> 
>> I'm having latest version of Freeswitch installed on Ubuntu 12.04.2 LTS  with the latest version of Openssl ('OpenSSL 1.0.1e 11 Feb 2013')
>> I'm using the default configuration and just uncommentated the ' <param name="ws-binding"  value=":5066"/> ' in internal.xml in order to have the support for webrtc. 
>> As the client I'm having JSSIP, the latest version with the adjustment to have (DtlsSrtpKeyAgreement:true). Both my clients and server located in the same network without having any firewalls therefore I'm not using any STUN server. I can make call but there is no sound! and after setting debig_level 10 and dial from extension 1003 to 5000 I'm having the following Alert in console and no sound:
>> 
>> [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!)
>> 
>> its the same case if I dial from extension 1002 to 1005! (Both are JSSIP clients) 
>> If I call from extension 1000 which is a SIP client set on my iPhone and call 1003 I'm having the following alerts but voice on both JSSIP client and SIP client:
>> 
>> 2013-08-05 14:17:05.222446 [ALERT] switch_rtp.c:4563 sofia/internal/1000 at 10.0.14.16:5060 timer while HOT
>> 2013-08-05 14:17:05.242449 [ALERT] switch_rtp.c:4546 sofia/internal/1000 at 10.0.14.16:5060 Hot Hit 1
>> 
>> And from extension 1003 (JSSIP) to extension 1000 ( SIP on Iphone) No voice and I'm getting the following alerts:
>> 
>> 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4546 sofia/internal/sip:1000 at 10.0.14.182:5065 Hot Hit 4
>> 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4563 sofia/internal/sip:1000 at 10.0.14.182:5065 timer while HOT
>> 2013-08-05 14:19:57.132441 [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!)
>> 
>> If I use bypass media or proxy media I will have voice on both JSSIP clients but cant ring any SIP clients and again no voice if I call 5000!
>> 
>> Wonder to know if there is any special setting required on FreeSwitch or its and issue from JSSIP? 
>> 
>> Thanks in advanced!
>> 
>> Sherry
> 
> 
> 
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