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@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@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@lists.freeswitch.org Reply-To: FreeSWITCH Users Help <freeswitch-users@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@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@lists.freeswitch.org Reply-To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Hi Michael,
I think I'll try sox, thanks.
Melanie
From: pablo platt <pablo.platt@gmail.com> Subject: [Freeswitch-users] Conference for Flash and WebRTC users Date: August 12, 2013 2:40:17 AM CDT To: freeswitch-users@lists.freeswitch.org Reply-To: FreeSWITCH Users Help <freeswitch-users@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@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@lists.freeswitch.org> Reply-To: FreeSWITCH Users Help <freeswitch-users@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@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@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@lists.freeswitch.org> Reply-To: FreeSWITCH Users Help <freeswitch-users@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@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@davri.com <mailto:bdfoster@davri.com> Indianapolis, Indiana
Sent from a mobile device.
On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev@c3i.bg <mailto:imitev@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@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@lists.freeswitch.org> Reply-To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Have you checked the logfile and netstat?
On 12 August 2013 09:51, Ashish Mishra <itsme.kunnu@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@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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From: Michael Jerris <mike@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@lists.freeswitch.org> Reply-To: FreeSWITCH Users Help <freeswitch-users@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@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@10.0.14.16:5060 timer while HOT 2013-08-05 14:17:05.242449 [ALERT] switch_rtp.c:4546 sofia/internal/1000@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@10.0.14.182:5065 Hot Hit 4 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4563 sofia/internal/sip:1000@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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