The alternative if you want in sync audio would be to use webrtc endpoints to listen. On Aug 14, 2013, at 7:19 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Icecast is not designed for low-latency and realtime audio. Its designed for higher quality reliable audio. The servers and the client libs both use latency and buffering to guarantee this. There are some techniques to reduce the buffering on at least the FS side but then you will start getting skips and resets if you miss any packets.
This comes up all the time when people first try doing this. Either you need to just accept the delay since most people will not even know its there or use some other method.
On Wed, Aug 14, 2013 at 5:12 PM, Stephen Dame <sdame@207me.com> wrote: Did you restart the freeswitch server after changes to the config files, I’m not sure 100% if reloadxml resets the conference settings.
Also I use some flash based players, I think there where client settiings for buffering. Can you set buffer in HTML tag?
Regards,
Stephen
207 Technology Group Inc. 1-888-229-9756 skype: Stephen_Dame
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jorge Núñez Sent: Wednesday, August 14, 2013 4:18 PM
To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Mod_shout using with icecast Big delay
Hi thanks, I set your configuration but nothing changed, I reduced the burst size and it takes me just 11s and also I tried with 8k, 16k and 32k and nothing change
<limits>
<clients>100</clients>
<sources>10</sources>
<threadpool>5</threadpool>
<queue-size>524288</queue-size>
<!-- <queue-size>524288</queue-size> -->
<client-timeout>30</client-timeout>
<header-timeout>15</header-timeout>
<source-timeout>10</source-timeout>
<!-- If enabled, this will provide a burst of data when a client
first connects, thereby significantly reducing the startup
time for listeners that do substantial buffering. However,
it also significantly increases latency between the source
client and listening client. For low-latency setups, you
might want to disable this. -->
<burst-on-connect>0</burst-on-connect>
<!-- same as burst-on-connect, but this allows for being more
specific on how much to burst. Most people won't need to
change from the default 64k. Applies to all mountpoints
-->
<burst-size>4096</burst-size>
</limits>
Regards
Jorge
2013/8/14 Stephen Dame <sdame@207me.com>
Jorge,
Play around with the burst size and queue size… here is my xml config around 3-4 second delay from 16k freeswitch conference.
To small a buffer and the players disconnect… Im also running icecast on same server.
<icecast>
<limits>
<clients>100</clients>
<sources>10</sources>
<threadpool>5</threadpool>
<queue-size>524288</queue-size>
<client-timeout>30</client-timeout>
<header-timeout>15</header-timeout>
<source-timeout>10</source-timeout>
<burst-on-connect>1</burst-on-connect>
<!-- same as burst-on-connect, but this allows for being more
specific on how much to burst. Most people won't need to
change from the default 64k. Applies to all mountpoints -->
<burst-size>65535</burst-size>
</limits>
Regards,
Stephen
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ken Rice Sent: Wednesday, August 14, 2013 12:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Mod_shout using with icecast Big delay
You have to keep in mind that icecast itself has a fair bit of buffering involved with it and theres not a lot you can do about that
On 8/14/13 11:30 AM, "Jorge Núñez" <jorgefren12@gmail.com> wrote:
Hi I am using mod_shout to send a conference call to icecast and hear it from a tag audio in html5 in realtime, but it has a big delay like 12 seconds. How can I reduce the latency of the audio sent from freeswitch or what can I do to improve this.
Regards
Jorge
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