<div dir="ltr">Thanks for your answer, well I changed this parameters on icecast.xml and the the delay reduce from 20s to 12s<div><br></div><div> <burst-on-connect>0</burst-on-connect></div><div> <burst-size>4096</burst-size></div>
<div><br></div><div><br></div><div>Well I was trying to reproduce mp3 and ogg but both have 12 s of delay. How can I reduce to maybe 1 or 2 seconds. </div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/8/7 <span dir="ltr"><<a href="mailto:icecast-request@xiph.org" target="_blank">icecast-request@xiph.org</a>></span><br>
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Today's Topics:<br>
<br>
1. Using freeswitch and Icecast (Jorge N??ez)<br>
2. Re: Using freeswitch and Icecast (Basil Mohamed Gohar)<br>
3. Re: Using freeswitch and Icecast (Thomas B. R?cker)<br>
4. Re: Using freeswitch and Icecast (TheDarkener)<br>
<br>
<br>
----------------------------------------------------------------------<br>
<br>
Message: 1<br>
Date: Tue, 6 Aug 2013 18:40:52 -0500<br>
From: Jorge N??ez <<a href="mailto:jorgefren12@gmail.com">jorgefren12@gmail.com</a>><br>
Subject: [Icecast] Using freeswitch and Icecast<br>
To: <a href="mailto:icecast@xiph.org">icecast@xiph.org</a><br>
Message-ID:<br>
<CAOP4xcztX5h4kJ-ftxYfaK=<a href="mailto:HQgqbPMQgy4FKZbc-VXBqHVTrUA@mail.gmail.com">HQgqbPMQgy4FKZbc-VXBqHVTrUA@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
Hi I am trying to use icecast to broadcast a realtime conference from<br>
freeswitch. But I am having a delay like 20 seconds then I reduced it to<br>
12s. But I don't know if somebody can help me how to reduce it as lower as<br>
possible.<br>
<br>
Thanks<br>
<br>
Jorge<br>
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Message: 2<br>
Date: Wed, 07 Aug 2013 02:48:16 -0400<br>
From: Basil Mohamed Gohar <<a href="mailto:basilgohar@librevideo.org">basilgohar@librevideo.org</a>><br>
Subject: Re: [Icecast] Using freeswitch and Icecast<br>
To: <a href="mailto:icecast@xiph.org">icecast@xiph.org</a><br>
Message-ID: <<a href="mailto:5201EDB0.9060301@librevideo.org">5201EDB0.9060301@librevideo.org</a>><br>
Content-Type: text/plain; charset=ISO-8859-1<br>
<br>
On 08/06/2013 07:40 PM, Jorge N??ez wrote:<br>
> Hi I am trying to use icecast to broadcast a realtime conference from<br>
> freeswitch. But I am having a delay like 20 seconds then I reduced it to<br>
> 12s. But I don't know if somebody can help me how to reduce it as lower<br>
> as possible.<br>
><br>
> Thanks<br>
><br>
> Jorge<br>
<br>
Jorge, first I'd like to know what you did to reduce the delay from 20<br>
to 12 seconds.<br>
<br>
Secondly, the delay can also be related to the following two issues:<br>
<br>
1. The audio codec being used<br>
2. The amount of buffering on the client side<br>
<br>
Using Ogg Vorbis for the audio, for example, will incur some delay, but<br>
that wouldn't explain 12 seconds. More than likely, a good portion of<br>
the delay is related to buffering on the client side.<br>
<br>
VLC media player allows you to set the buffering on the client side, you<br>
may consider trying that and experimenting with different settings to<br>
see how little delay you can get by reducing buffering. But keep in<br>
mind, a smaller buffer exposes your client to more network-related<br>
transmission errors, which may result in Icecast dropping your client<br>
altogether or the sound coming through with stuttering.<br>
<br>
The fact of the matter is that while Icecast can be used for<br>
broadcasting live audio, it's a very simple protocol that does not have<br>
any mechanism to ensure that all clients are receiving the audio at the<br>
same time. The audio is being broadcast as basically an HTTP download<br>
without a Content-Length header. It's up to the client (i.e.,<br>
application) to present it to the user with or without delay, depending<br>
on configuration as I mentioned above.<br>
<br>
--<br>
Libre Video<br>
<a href="http://librevideo.org" target="_blank">http://librevideo.org</a><br>
<br>
<br>
------------------------------<br>
<br>
Message: 3<br>
Date: Wed, 07 Aug 2013 07:06:23 +0000<br>
From: "Thomas B. R?cker" <<a href="mailto:thomas@ruecker.fi">thomas@ruecker.fi</a>><br>
Subject: Re: [Icecast] Using freeswitch and Icecast<br>
To: <a href="mailto:icecast@xiph.org">icecast@xiph.org</a><br>
Message-ID: <<a href="mailto:5201F1EF.5030204@ruecker.fi">5201F1EF.5030204@ruecker.fi</a>><br>
Content-Type: text/plain; charset=ISO-8859-1<br>
<br>
what-he-said<br>
<br>
On 08/07/2013 06:48 AM, Basil Mohamed Gohar wrote:<br>
> On 08/06/2013 07:40 PM, Jorge N??ez wrote:<br>
>> Hi I am trying to use icecast to broadcast a realtime conference from<br>
>> freeswitch. But I am having a delay like 20 seconds then I reduced it to<br>
>> 12s. But I don't know if somebody can help me how to reduce it as lower<br>
>> as possible.<br>
>><br>
>> Thanks<br>
>><br>
>> Jorge<br>
> Jorge, first I'd like to know what you did to reduce the delay from 20<br>
> to 12 seconds.<br>
><br>
> Secondly, the delay can also be related to the following two issues:<br>
><br>
> 1. The audio codec being used<br>
> 2. The amount of buffering on the client side<br>
><br>
> Using Ogg Vorbis for the audio, for example, will incur some delay, but<br>
> that wouldn't explain 12 seconds. More than likely, a good portion of<br>
> the delay is related to buffering on the client side.<br>
><br>
> VLC media player allows you to set the buffering on the client side, you<br>
> may consider trying that and experimenting with different settings to<br>
> see how little delay you can get by reducing buffering. But keep in<br>
> mind, a smaller buffer exposes your client to more network-related<br>
> transmission errors, which may result in Icecast dropping your client<br>
> altogether or the sound coming through with stuttering.<br>
><br>
> The fact of the matter is that while Icecast can be used for<br>
> broadcasting live audio, it's a very simple protocol that does not have<br>
> any mechanism to ensure that all clients are receiving the audio at the<br>
> same time. The audio is being broadcast as basically an HTTP download<br>
> without a Content-Length header. It's up to the client (i.e.,<br>
> application) to present it to the user with or without delay, depending<br>
> on configuration as I mentioned above.<br>
<br>
In addition, why would you require the broadcast to be as low latency as<br>
the conversation itself?<br>
Even professional broadcasting will have a delay of anywhere between a<br>
couple hundred miliseconds and several seconds.<br>
<br>
If you need real time interaction or want to have people dial in, then<br>
those people should use the VoIP bidirectional channel and not the<br>
broadcast. That's how call-in shows on the radio do it too.<br>
<br>
Also while you can get the latency into the 1-2s range with Icecast,<br>
this will come at the cost of stability and increased configuration<br>
complexity across the whole chain, including the listener (especially<br>
the listener side software, as those tend to have the largest buffers).<br>
Icecast is simply not a low latency solution, let alone real-time. If<br>
you are looking for sub-second latency, Icecast is the wrong tool or you<br>
need to accommodate how broadcasting works in your workflow.<br>
<br>
Cheers<br>
<br>
Thomas<br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 4<br>
Date: Wed, 07 Aug 2013 09:06:00 -0700<br>
From: TheDarkener <<a href="mailto:thedarkener@logicalnetworking.net">thedarkener@logicalnetworking.net</a>><br>
Subject: Re: [Icecast] Using freeswitch and Icecast<br>
To: <a href="mailto:Icecast@xiph.org">Icecast@xiph.org</a><br>
Message-ID: <<a href="mailto:52027068.7070802@logicalnetworking.net">52027068.7070802@logicalnetworking.net</a>><br>
Content-Type: text/plain; charset=ISO-8859-1<br>
<br>
All of that being said, using ogg/opus seems like it should be more<br>
geared toward what you're trying to do - though what everyone above has<br>
said still stands, regardless of codec.<br>
<br>
<br>
On 08/07/2013 12:06 AM, "Thomas B. R?cker" wrote:<br>
> what-he-said<br>
><br>
> On 08/07/2013 06:48 AM, Basil Mohamed Gohar wrote:<br>
>> On 08/06/2013 07:40 PM, Jorge N??ez wrote:<br>
>>> Hi I am trying to use icecast to broadcast a realtime conference from<br>
>>> freeswitch. But I am having a delay like 20 seconds then I reduced it to<br>
>>> 12s. But I don't know if somebody can help me how to reduce it as lower<br>
>>> as possible.<br>
>>><br>
>>> Thanks<br>
>>><br>
>>> Jorge<br>
>> Jorge, first I'd like to know what you did to reduce the delay from 20<br>
>> to 12 seconds.<br>
>><br>
>> Secondly, the delay can also be related to the following two issues:<br>
>><br>
>> 1. The audio codec being used<br>
>> 2. The amount of buffering on the client side<br>
>><br>
>> Using Ogg Vorbis for the audio, for example, will incur some delay, but<br>
>> that wouldn't explain 12 seconds. More than likely, a good portion of<br>
>> the delay is related to buffering on the client side.<br>
>><br>
>> VLC media player allows you to set the buffering on the client side, you<br>
>> may consider trying that and experimenting with different settings to<br>
>> see how little delay you can get by reducing buffering. But keep in<br>
>> mind, a smaller buffer exposes your client to more network-related<br>
>> transmission errors, which may result in Icecast dropping your client<br>
>> altogether or the sound coming through with stuttering.<br>
>><br>
>> The fact of the matter is that while Icecast can be used for<br>
>> broadcasting live audio, it's a very simple protocol that does not have<br>
>> any mechanism to ensure that all clients are receiving the audio at the<br>
>> same time. The audio is being broadcast as basically an HTTP download<br>
>> without a Content-Length header. It's up to the client (i.e.,<br>
>> application) to present it to the user with or without delay, depending<br>
>> on configuration as I mentioned above.<br>
> In addition, why would you require the broadcast to be as low latency as<br>
> the conversation itself?<br>
> Even professional broadcasting will have a delay of anywhere between a<br>
> couple hundred miliseconds and several seconds.<br>
><br>
> If you need real time interaction or want to have people dial in, then<br>
> those people should use the VoIP bidirectional channel and not the<br>
> broadcast. That's how call-in shows on the radio do it too.<br>
><br>
> Also while you can get the latency into the 1-2s range with Icecast,<br>
> this will come at the cost of stability and increased configuration<br>
> complexity across the whole chain, including the listener (especially<br>
> the listener side software, as those tend to have the largest buffers).<br>
> Icecast is simply not a low latency solution, let alone real-time. If<br>
> you are looking for sub-second latency, Icecast is the wrong tool or you<br>
> need to accommodate how broadcasting works in your workflow.<br>
><br>
> Cheers<br>
><br>
> Thomas<br>
><br>
> _______________________________________________<br>
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<br>
--<br>
Jordan (PGP: 0xDA470FF8)<br>
<br>
<br>
<br>
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End of Icecast Digest, Vol 111, Issue 5<br>
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</blockquote></div><br></div></div>