[Freeswitch-users] Strange media behavior on WebRTC
Michael Jerris
mike at jerris.com
Mon Jun 16 19:42:00 MSD 2014
use ldd?
On Jun 15, 2014, at 3:09 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> And looking closer at Brian's Makefile it seems that my CentOS system already used OpenSSL 1.0.1g.
>
> Let me know if there is a way to verify the OpenSSL version FreeSWITCH uses.
>
> Otherwise, I will see what my OS options are.
>
> Thanks for all your help and I want to make sure it's clear that this is a small problem compared to the amazing functionality FreeSWITCH provides.
>
>
> On Sat, Jun 14, 2014 at 8:01 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> They all have different OpenSSL versions but they all have the latest available for them.
>
> I tried building FreeSWITCH on my CentOS 5 with the latest OpenSSL using these instructions:
> http://wiki.freeswitch.org/wiki/Installation_Guide#FreeSWITCH_v.1.4.2Fmaster_openssl_requirement_change
>
> I had to fudge it a little to make it work with Brian's Makefile.
>
> The problem is still there but I am not sure switching to the new OpenSSL worked. The active version of OpenSSL on the OS is still old.
>
> Is there a way to check which OpenSSL version FreeSWITCH is using?
>
>
>
> On Sat, Jun 14, 2014 at 2:15 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> My only suggestion is check the version of openssl?
>
>
> On Sat, Jun 14, 2014 at 12:26 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> Thanks Anthony, I did that. No response yet.
>
> However, I did get a VM with Debian 7, and indeed this problem does not happen there.
>
> I am not sure where to go from here. I am kind of stuck with CentOS as my production system at least for the next several months.
>
>
> On Thu, Jun 12, 2014 at 11:51 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> You can email consulting at freeswitch.org and see if we can find a way.
>
>
> On Thu, Jun 12, 2014 at 1:28 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> I have not tried it on the dev channel but it's happening on both stable and is worse on Beta. It is not happening on your site even with the same stable or beta browser, so it has to be something different with my installation.
>
> You are absolutely right - the first thing to do is reproduce it. I understand that this may take some time and perhaps getting on the phone together, that's why I am asking for commercial support. I don't expect guaranteed results of course - just need some sanity check :-)
>
>
> On Thu, Jun 12, 2014 at 11:14 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> I'm using 37 (dev channel).
> I am not sure what we can do to fix the problem if its not something we can reproduce.
> Have you tried it on canary or dev channel?
>
>
> On Thu, Jun 12, 2014 at 12:21 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> Thanks, this is identical to what I have. The problem for me is getting worse on Chrome 36 beta. On 35 or earlier I was at least able to find a delay setting where the symptoms were minimized. Now on 36 it's always bad.
>
> I have an out of the box FreeSWITCH installation on Windows where all I did was uncomment in the ws-binding in internal.xml sip profile and added this extension to public.xml dial plan as the very first extension. Then I changed the prefs on http://webrtc.freeswitch.org/sipjs/index.html#8765 to point to this FreeSWITCH.
>
> Could I get some commercial support for this? Should I email consulting at freeswitch.com? I'd be happy to give you access to my installations where this is happening.
>
>
> On Thu, Jun 12, 2014 at 9:55 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> <extension name="8765">
>
> <condition field="destination_number" expression="^8765$">
>
> <action application="set" data="answer_delay=3000"/>
>
> <action application="answer"/>
>
> <action application="playback" data="tone_stream://%(1000,0,600)" />
>
> <action application="conference" data="$1 at default"/>
>
> </condition>
>
> </extension>
>
>
>
> On Thu, Jun 12, 2014 at 11:47 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> Hi Anthony,
>
> could you send me the XML for the extension you used to set up this example?
>
>
>
> On Sun, Jun 8, 2014 at 7:11 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> Hey Anthony,
>
> just in case I am missing something obvious, could you send me the extension in the dialplan you set up with my example?
>
>
> On Sun, Jun 8, 2014 at 6:59 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> Yeah, I saw sipjs vs jssp but I pointed my own jssip client to your example and there was no noise, so it's not the client. I also am setting my stun servers array to empty to speed up the connection but that also does not seem to matter. I'll keep digging.
>
>
> On Sun, Jun 8, 2014 at 6:15 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> Also it's using sipjs.com not jssip.
>
> On Jun 8, 2014 8:14 AM, "Anthony Minessale" <anthony.minessale at gmail.com> wrote:
> Debian 7 on proxmox vm.
> On Jun 6, 2014 10:18 PM, "Oleg Stolyar" <olegstolyar at gmail.com> wrote:
> Anthony,
>
> Another update - I thought that the only thing my original installation and the Debian VM had in common was that both were VMs. So I went ahead and installed FreeSWITCH on my local Windows machine. Same problem is happening here as well.
>
> I am extremely curious what's different between your installation and my three setups (CentOS, Debian and Windows) that accounts for the difference in this behavior. Could you tell me what machine/OS you are using?
>
>
> On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> Hi Anthony,
>
> I dug further and I can reproduce the issue on a completely different VM (Debian on Windows host). I installed the latest FS master and even switched to the vanilla conf to eliminate the possibility of my config causing this issue.
>
> It's still happening. If you have time to look into this some more, I'll be happy to upload my whole VM for someone to take a look at.
>
>
> On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> Yep it's the same.
>
> Thank you
> Oleg
>
> On Jun 4, 2014 4:05 PM, "Anthony Minessale" <anthony.minessale at gmail.com> wrote:
> What if you play an audio file? is it the same?
> I can only guess really.
>
>
>
> On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> I tried this:
> <action application="set" data="timer_name=soft"/>
> <action application="playback" data="tone_stream://%(1000,0,600)" />
>
> Didn't help unfortunately (unless this is the wrong syntax for setting the variable). Completely agreed on the AWS instances - in fact we started on an older m1 instance and the voice quality was pretty bad and there were frequent delays. Then we switch to an m3 and it was much better.
>
> Any advice on setting enter-sound per member? Or setting a no-enter-sound flag - similar to the nomoh flag?
>
>
>
> On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> Try setting the channel var timer_name=soft before playing the beep. It may be jittery media flow causing your problem.
> My only advice on amazon is use the more expensive one because the lower cost ones are not reliable for cpu time.
>
>
>
> On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> OK, thanks. It's still happening with the latest master, so I have to assume that this is something to do with the AWS VMs. It would not be the first time strange things happen there.
>
> I was trying to work around the problem and noticed that if instead of playing the beep using playback app before placing the user into the conference, I make the beep the enter-sound of the conference, the problem does not happen.
>
> However, I need to only play this sound for users joining the conference from one dialplan but not from another. I cannot find a way to do it. I looked at
> <action application="set" data="conference_enter_sound=silence_stream://10"/>
> but it changes the enter sound for the conference for everyone, so anyone entering after the user who set this up will hear the same sound.
>
> Is there a way to assign conference params per user (like member flags)?
>
>
> On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> If you opt to use master, ALWAYS use latest master since that is the nature of master. I have no idea if your problem is related to those things but the server I pointed you to test is running basically latest master.
>
>
>
> On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> Thanks Anthony!
>
> I tried adding this var and also tried uncommenting the suppress_cng line in my sip profile. It did not seem to make a difference. I pointed my client code to your sample and it worked fine - no noise, so it's not a problem with the client side.
>
> Do I need to upgrade to the latest master? I am using master from 5/22 version 1.5.13b
>
> Would it help if gave you my test ws server and sip address so you could reproduce this?
>
> Is it possible that the problem is with my AWS instance or CentOS 5.9?
>
>
>
>
> On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> I set up your test ext here:
> https://webrtc.freeswitch.org/sipjs/index.html#8765
>
> And it seems to work.
>
> Try this in vars.xml
>
> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/>
>
> I don't think the silence suppression works very well in the browser.
>
>
>
>
>
>
>
> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> In case it was not clear from the previous email - this only happens with conferences. If after the beep, instead of placing the call into a conference, I just play some music, there are no problems.
>
>
> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> Hi guys,
>
> I am connecting to FreeSWITCH using WebRTC and after a user calls in, I play a beep and put them into a conference. Two strange things happen after that:
>
> 1. There is a noise in the user's leg that someone described like a wind in the tunnel. The noise lasts for several seconds, then the user can hear the hold music. The noise does not seem to happen if I don't play the beep before the conference.
>
> 2. About 3 seconds into the call, there is another very short beep - like a small portion of the original beep that the user hears.
>
> This absolutely does not happen with softphones - only with WebRTC. I tried it with Chrome and Opera - same thing is happening on both.
>
> I tried increasing the answer delay from 2 seconds to 3 and the noise now happens more rarely but the extra beep is still there almost every time. I am using JsSIP 3.0 but since it only handles signalling I don't think the problem is with the JS library.
>
> Has anyone run into this? Any advice?
>
> Here is the relevant excerpt from my dialplan.
>
> <condition field="destination_number" expression="^conf-(\S+)$">
> <action application="set" data="answer_delay=3000"/>
> <action application="answer"/>
> <action application="playback" data="tone_stream://%(1000,0,600)" />
> <action application="conference" data="$1 at default"/>
> </condition>
>
>
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