You can also use park to connect two users, and use eavesdrop when you need to connect the third one

On Oct 7, 2015 9:56 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
Basically I am willing to burn some extra electricity for the ability to have other people easily join the call, automatically record when a second person joins and other features I get from conference for free (or at least at the cost of extra electricity)

Besides, all my calls from both internal and external users are incoming, so if I did not use conference, I would have to use park and I am not sure that a common codec could be negotiated in this case.



On Wed, Oct 7, 2015 at 12:31 PM, Michael Jerris <mike@jerris.com> wrote:
If you only have 2 calls, why not just bridge them instead of having all the extra overhead of a conference?  Also, if you have only 2 people, one is wideband and the other narrow, you get no benefit other than you get to create a bit more extra heat and use some more electricity.  If this was a bridged call, you could negotiate common codecs.

On Oct 7, 2015, at 2:12 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:

Yes, it's almost always one internal and one external user.

On Wed, Oct 7, 2015 at 12:00 PM, Michael Jerris <mike@jerris.com> wrote:
I think down is a bit cheaper cpu than up, but also, do you have the same number of internal and external users?

On Oct 7, 2015, at 11:09 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:

Interestingly, when I switched my conferences from 8000 to 16000 sample rate and left the codecs at G722 for internal users and G711 for external, the CPU usage went down by as much as 15%.

Is there an explanation for this?

On Wed, Oct 7, 2015 at 7:53 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks guys!

On Wed, Oct 7, 2015 at 5:58 AM, Steven Ayre <steveayre@gmail.com> wrote:
http://tools.ietf.org/html/rfc3551
   Even though the actual sampling rate for G.722 audio is 16,000 Hz,
   the RTP clock rate for the G722 payload format is 8,000 Hz because
   that value was erroneously assigned in RFC 1890 and must remain
   unchanged for backward compatibility.  The octet rate or sample-pair
   rate is 8,000 Hz.

On 7 October 2015 at 00:01, Michael Jerris <mike@jerris.com> wrote:
G722/8000 is actually 16k not 8, an accident of history...


On Tuesday, October 6, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Yeah but Chrome seems to only be offering G722 at 8000
a=rtpmap:9 G722/8000

Am I reading this correctly?

On Tue, Oct 6, 2015 at 3:32 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
G722 is actually 16khz which can pass for hd
isac will do higher sample rates as well like 16 and 32 but it will probably also be cpu intensive.


On Tue, Oct 6, 2015 at 5:17 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:

With browsers/webrtc I believe your choice is only between opus and g711

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18

On Oct 7, 2015 12:04 AM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
My endpoint is Chrome/WebRTC.  Here is what the current version sends in the the INVITE. 

Doesn't look like G722/16000 is there.  How is ISAC in terms of CPU consumption?

a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000


On Tue, Oct 6, 2015 at 1:49 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
If your endpoint supports it, G722@16000h gives a very decent voice quality. SILK also supports 16KHz.

here's some information related to codec complexity:
https://txlab.wordpress.com/2015/04/10/testing-freeswitch-performance-on-scaleway-c1/



On Tue, Oct 6, 2015 at 10:40 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks guys!

Anthony, using your suggestion means that I won't have HD voice, correct?  HD voice is the primary reason I am considering OPUS instead of G.722.

Is there some codec/setting that will let me have HD voice for less CPU and perhaps more bandwidth?

It's not a must - I can scale up if needed, just wondering.

On Tue, Oct 6, 2015 at 1:35 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
typo: opus@8000h@20i

On Tue, Oct 6, 2015 at 3:33 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
OPUS is an expensive codec, so any conversion between OPUS and
uncompressed audio is CPU-consuming. I don;t think that resampling
would play much of a role in CPU consumption.

On Tue, Oct 6, 2015 at 10:29 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
> Hi guys,
>
> I have a bunch of conferences on my FS.  Users are either G.711/8000 or
> G.722/8000.  Conference sample rate is set to 8000.
>
> When I switched some users to OPUS instead, the CPU consumption jumped 3-4
> times even though the conference rate is still 8000.  Is this expected?
>
> I was hoping to be able to save some CPU by setting the conference sample
> rate to auto (so it only used high-quality sampling for users that can
> support it)  but it seems it won't work.  Any ideas?
>
> _________________________________________________________________________
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_________________________________________________________________________
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--
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬






--
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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