<html><head><meta http-equiv="Content-Type" content="text/html charset=windows-1252"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div>Julien,</div><div><br></div><div>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. </div><div><br></div><div>- Chris</div><br><div><div>On Aug 12, 2013, at 9:46 AM, <a href="mailto:freeswitch-users-request@lists.freeswitch.org">freeswitch-users-request@lists.freeswitch.org</a> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">Send FreeSWITCH-users mailing list submissions to<br><span class="Apple-tab-span" style="white-space:pre"> </span><a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br><br>To subscribe or unsubscribe via the World Wide Web, visit<br><span class="Apple-tab-span" style="white-space:pre"> </span>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>or, via email, send a message with subject or body 'help' to<br><span class="Apple-tab-span" style="white-space:pre"> </span>freeswitch-users-request@lists.freeswitch.org<br><br>You can reach the person managing the list at<br><span class="Apple-tab-span" style="white-space:pre"> </span>freeswitch-users-owner@lists.freeswitch.org<br><br>When replying, please edit your Subject line so it is more specific<br>than "Re: Contents of FreeSWITCH-users digest..."<br>Today's Topics:<br><br> 1. How to create a single outbound ESL socket at<span class="Apple-tab-span" style="white-space:pre"> </span>freeswitch<br> startup. (julien terrasson)<br> 2. uuid_displace: change loudness level of audio<br> (Melanie Treitinger)<br> 3. Conference for Flash and WebRTC users (pablo platt)<br> 4. Re: Error in launching fs_cli (Ashish Mishra)<br> 5. Re: garbled audio with G726-32,<span class="Apple-tab-span" style="white-space:pre"> </span>other codecs are fine<br> (Michael Jerris)<br> 6. Re: Error in launching fs_cli (Steven Ayre)<br> 7. Re: Issue with JSSIP + Freeswitch (Michael Jerris)<br><br><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>From: </b></span><span style="font-family:'Helvetica'; font-size:medium;">julien terrasson <julien.terrasson@gmail.com><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Subject: </b></span><span style="font-family:'Helvetica'; font-size:medium;"><b>[Freeswitch-users] How to create a single outbound ESL socket at freeswitch startup.</b><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Date: </b></span><span style="font-family:'Helvetica'; font-size:medium;">August 12, 2013 2:01:37 AM CDT<br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">freeswitch-users@lists.freeswitch.org<br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Reply-To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br></span></div><br><br><div dir="ltr"><div><div><div>Hi,<br><br></div>I would like to create a single outbound ESL socket that would establish the connection with the ESL server at freeswitch startup.<br>This connexion would remains up and would used to transfer events from calls going through this server.<br>
</div>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).<br></div><div>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 ?)<br>
<br></div><div>Hope somebody can help..<br><br></div><div>Regards,<br><br></div><div>Julien<br></div><div><br><br></div><div><br></div><br><div><br></div></div>
<br><br><br><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>From: </b></span><span style="font-family:'Helvetica'; font-size:medium;">Melanie Treitinger <treitinger@as-infodienste.de><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Subject: </b></span><span style="font-family:'Helvetica'; font-size:medium;"><b>[Freeswitch-users] uuid_displace: change loudness level of audio</b><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Date: </b></span><span style="font-family:'Helvetica'; font-size:medium;">August 12, 2013 3:33:36 AM CDT<br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">freeswitch-users@lists.freeswitch.org<br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Reply-To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br></span></div><br><br>
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<font face="Arial">Hi Michael,<br>
<br>
I think I'll try sox, thanks.<br>
<br>
<br>
Melanie</font>
</div>
<br><br><br><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>From: </b></span><span style="font-family:'Helvetica'; font-size:medium;">pablo platt <pablo.platt@gmail.com><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Subject: </b></span><span style="font-family:'Helvetica'; font-size:medium;"><b>[Freeswitch-users] Conference for Flash and WebRTC users</b><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Date: </b></span><span style="font-family:'Helvetica'; font-size:medium;">August 12, 2013 2:40:17 AM CDT<br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">freeswitch-users@lists.freeswitch.org<br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Reply-To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br></span></div><br><br><div dir="ltr"><div><div><div>Hi,<br><br></div>I'm currently using a RTMP server for audio conference.<br></div>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.<br>
<br></div><div>I'm trying to evaluate FreeSWITCH so I'll be able to support both Flash and WebRTC users in a conference.<br><br></div><div>My use case is two types of conferences:<br></div><div>- A conference with up to 5 participants all speaking.<br>
</div><div>- A conference with 20-30 participants with 2-3 speakers.<br><br><div>What part require more CPU?<br></div><div>- Encryption (DTLS-SRTP)<br></div><div>- Transcoding<br></div>- Mixing<br><br></div><div>Does FS mix a channel separately for each participant or can it reuse the same mix?<br>
</div><div>For example, if I have 1 speaker and 20 listeners, 10 with speex and 10 with opus, does FS produce 2 streams or 20?<br><br></div><div>Does FS exclude silent participants from the mix to save CPU?<br></div><div>
<br></div><div>Thanks<br></div><div><br></div><div><br><br></div></div>
<br><br><br><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>From: </b></span><span style="font-family:'Helvetica'; font-size:medium;">Ashish Mishra <itsme.kunnu@gmail.com><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Subject: </b></span><span style="font-family:'Helvetica'; font-size:medium;"><b>Re: [Freeswitch-users] Error in launching fs_cli</b><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Date: </b></span><span style="font-family:'Helvetica'; font-size:medium;">August 12, 2013 3:51:45 AM CDT<br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Reply-To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br></span></div><br><br><p>I reinstalled freeswitch but still i am getting the same error :<br>
fs_cli .c:1455 main() Error Connecting [Socket Connection Error ] mod_socket is also loaded...i have also flushed the firewall rules...pls help</p>
<div class="gmail_quote">On Aug 6, 2013 2:30 AM, "Ashish Mishra" <<a href="mailto:itsme.kunnu@gmail.com">itsme.kunnu@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p>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 :<br>
fs_cli .c:1455 main() Error Connecting [Socket Connection Error ] <br>
</p>
</blockquote></div>
<br><br><br><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>From: </b></span><span style="font-family:'Helvetica'; font-size:medium;">Michael Jerris <mike@jerris.com><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Subject: </b></span><span style="font-family:'Helvetica'; font-size:medium;"><b>Re: [Freeswitch-users] garbled audio with G726-32, other codecs are fine</b><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Date: </b></span><span style="font-family:'Helvetica'; font-size:medium;">August 12, 2013 9:14:45 AM CDT<br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Reply-To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br></span></div><br><br>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.<br><br>Mike<br><br>On Aug 4, 2013, at 1:53 AM, Ivan Mitev <imitev@c3i.bg> wrote:<br><br><blockquote type="cite">Thanks for the suggestion but I'm testing with G726-32, not AAL2-G726-32 <br>; so bitpacking shouldn't be used. By the way when I tested with AAL2 to <br>the linphone client I only got cracks and whitenoise, I've forgot to <br>mention that in my post.<br><br>That said I've tried to uncomment and set <param name="bitpacking" <br>value="none"/> in internal.xml ("none" is a wild guess - I couldn't find <br>any doc on values accepted by this parameter), but that doesn't help.<br><br>Speex: the ATAs don't support it. And being stubborn I'd like to <br>understand what's wrong with G726 :)<br><br><br>On 08/04/2013 05:22 AM, Jeff Leung wrote:<br><blockquote type="cite"><br>You can turn off G726 AAC bit-packing in spandsp.conf.xml.<br><br>By the way, there are other codecs out there you can try. SPEEX comes <br>to mind if all your endpoints don’t deal with the PSTN.<br><br>*From:*freeswitch-users-bounces@lists.freeswitch.org <br>[mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of <br>*Brian Foster<br>*Sent:* Saturday, August 3, 2013 2:37 PM<br>*To:* FreeSWITCH Users Help<br>*Subject:* Re: [Freeswitch-users] garbled audio with G726-32, other <br>codecs are fine<br><br>AAC bitpacking by any chance? I thought I had a similar issue, <br>happened so long ago I cant remember what I did.<br><br>Thank you,<br><br>Brian Foster<br>Project Manager/Owner's Rep.<br>Davri Investments, Inc.<br>O: 317-787-2686 x2102<br>M: 317-600-9753<br>E: bdfoster@davri.com <mailto:bdfoster@davri.com><br>Indianapolis, Indiana<br><br>Sent from a mobile device.<br><br>On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev@c3i.bg <br><mailto:imitev@c3i.bg>> wrote:<br><br>Hello<br><br>I'm migrating an office setup from asterisk to FS and in the process I<br>was considering using G726-32 for some bandwidth starved remote<br>endpoints. However I only get metallic/garbled audio with that codec<br>even when simply playing moh to the endpoint, while other codecs work<br>fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds<br>marginally better but still garbled and really worse than G711.<br><br>The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest<br>(centos6 64bit host). But please don't shoot ! :) - I know about virtual<br>environment limitations but for these tests the host is only lightly<br>loaded, there aren't any calls to the FS instance except my tests, and<br>the fact that it works with other codecs makes me think that<br>virtualization is not the issue here. I may be wrong though.<br><br>Is there any guide for debugging that kind of problem before reverting<br>to a fresh install on bare-metal with the latest HEAD ? Until now I've<br>tried:<br><br>- improving timers ; but the default soft timer (which I guess uses<br>timerd) works best. The time interval between sent packets on a tcpdump<br>trace looks identical to the output of "timer_test", so that doesn't<br>seem to be a network/jitter problem. And there's no problem with other<br>codecs, but maybe G726-XX is specific. For info the guest's clocksource<br>is kvm_clock, while the host uses tsc.<br><br>- using different endpoints: the production ones are Linksys PAP2<br>("fixed" for 20ms psize, and G726-32 SDP type indentification), but the<br>same thing happens with linphone on a fedora 19 laptop.<br><br>A call with rtp media going through FS without transcoding - G726-32 to<br>G726-32 - works perfectly (I can't hear the difference with G711). The<br>problem is only when there's transcoding to G726 (from wav for moh, or<br>from any other codec when bridging). I've looked at the wiki, posts,<br>changelogs, jira, ..., but am a bit at a loss now.<br><br>Any pointers ?<br><br>Except that little problem, FS rocks, and I'm happy I can finally ditch<br>asterisk. Kudos to the core devs and contributors.<br><br>Ivan<br><br></blockquote></blockquote><br><br><br><br><br><br><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>From: </b></span><span style="font-family:'Helvetica'; font-size:medium;">Steven Ayre <steveayre@gmail.com><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Subject: </b></span><span style="font-family:'Helvetica'; font-size:medium;"><b>Re: [Freeswitch-users] Error in launching fs_cli</b><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Date: </b></span><span style="font-family:'Helvetica'; font-size:medium;">August 12, 2013 9:44:38 AM CDT<br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Reply-To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br></span></div><br><br>Have you checked the logfile and netstat?<div><br><br><div class="gmail_quote">On 12 August 2013 09:51, Ashish Mishra <span dir="ltr"><<a href="mailto:itsme.kunnu@gmail.com" target="_blank">itsme.kunnu@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p>I reinstalled freeswitch but still i am getting the same error :<br>
fs_cli .c:1455 main() Error Connecting [Socket Connection Error ] mod_socket is also loaded...i have also flushed the firewall rules...pls help</p><div class="HOEnZb"><div class="h5">
<div class="gmail_quote">On Aug 6, 2013 2:30 AM, "Ashish Mishra" <<a href="mailto:itsme.kunnu@gmail.com" target="_blank">itsme.kunnu@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p>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 :<br>
fs_cli .c:1455 main() Error Connecting [Socket Connection Error ] <br>
</p>
</blockquote></div>
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<br><br><br><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>From: </b></span><span style="font-family:'Helvetica'; font-size:medium;">Michael Jerris <mike@jerris.com><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Subject: </b></span><span style="font-family:'Helvetica'; font-size:medium;"><b>Re: [Freeswitch-users] Issue with JSSIP + Freeswitch</b><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Date: </b></span><span style="font-family:'Helvetica'; font-size:medium;">August 12, 2013 9:45:46 AM CDT<br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Reply-To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br></span></div><br><br><meta http-equiv="Content-Type" content="text/html charset=us-ascii"><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">We only use non rfc-1918 ip's by default. If you want to use 1918 ip's you need to tweak acls<div><br></div><div>Mike</div><div><br></div><div><br><div><div>On Aug 5, 2013, at 8:24 AM, Shahrzad A. <<a href="mailto:shahrzad.aziminia@gmail.com">shahrzad.aziminia@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div dir="ltr">Hi everyone<div><br></div><div>I'm having latest version of Freeswitch installed on Ubuntu 12.04.2 LTS with the latest version of Openssl (<font face="arial, sans-serif">'OpenSSL 1.0.1e 11 Feb 2013')</font><br>
<font face="arial, sans-serif">I'm using the default configuration and just uncommentated the '</font> <param name="ws-binding" value=":5066"/> ' in internal.xml in order to have the support for webrtc. </div>
<div>As the client I'm having JSSIP, the latest version with the adjustment to have (<font face="Lucida Grande, sans-serif">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:</font></div>
<div><span style="font-family: 'Lucida Grande', sans-serif; font-size: 13px; "><div><br></div><div>[ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!)</div><div><br></div></span></div>
<div><font face="Lucida Grande, sans-serif">its the same case if I dial from extension 1002 to 1005! (Both are JSSIP clients) </font></div><div><font face="Lucida Grande, sans-serif">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:</font></div>
<div><font face="Lucida Grande, sans-serif"><div><br></div><div>2013-08-05 14:17:05.222446 [ALERT] switch_rtp.c:4563 sofia/internal/<a href="http://1000@10.0.14.16:5060/">1000@10.0.14.16:5060</a> timer while HOT</div>
<div>2013-08-05 14:17:05.242449 [ALERT] switch_rtp.c:4546 sofia/internal/<a href="http://1000@10.0.14.16:5060/">1000@10.0.14.16:5060</a> Hot Hit 1</div><div><br></div><div>And from extension 1003 (JSSIP) to extension 1000 ( SIP on Iphone) No voice and I'm getting the following alerts:</div>
<div><div><br></div><div>2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4546 sofia/internal/<a href="http://sip:1000@10.0.14.182:5065/">sip:1000@10.0.14.182:5065</a> Hot Hit 4</div><div>2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4563 sofia/internal/<a href="http://sip:1000@10.0.14.182:5065/">sip:1000@10.0.14.182:5065</a> timer while HOT</div>
<div>2013-08-05 14:19:57.132441 [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!)</div></div><div><br></div><div>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!</div>
<div><br></div><div>Wonder to know if there is any special setting required on FreeSwitch or its and issue from JSSIP? </div><div><br></div><div>Thanks in advanced!</div><div><br></div><div>Sherry</div></font></div></div></blockquote></div><br></div></div><br><br>_______________________________________________<br>FreeSWITCH-users mailing list<br>FreeSWITCH-users@lists.freeswitch.org<br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br></blockquote></div><br></body></html>