No, it just disables the proper behavior of our rtp stack which is to have time still pass when not sending RTP. With the param enabled, the timestamps only go up when we are actually writing packets which is incorrect but does fix your problem ;) On Tue, Aug 5, 2014 at 12:48 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Tony!
Will there be any side effects if I use this setting?
Oleg On Aug 5, 2014 12:06 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
Looks like if you add
<action application="set" data= "rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
before you call answer or the conference it fixes it.
So the media engine is getting confused by our timestamps at the beginning of the call.
On Tue, Jun 17, 2014 at 12:27 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael! I've been testing 12000 and 16000 for a couple of days now. So far 12000 seems to use marginally (very marginally) less CPU than 16000. I'll monitor CPU for several more days before settling on one or the other.
On Tue, Jun 17, 2014 at 7:27 AM, Michael Jerris <mike@jerris.com> wrote:
if you have to up the rate to get around this issue, setting it to 16k instead of 12k when your endpoints are all 8k should use less cpu.
On Jun 17, 2014, at 1:26 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
For what it's worth, opus usually costs about the same bandwidth of ulaw. If you are using pstn callers anyway ulaw is fine. On Jun 16, 2014 11:23 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
Totally agree about the desire to figure it out. If we can do it, that would great.
PCMU vs OPUS is a separate issue from the noise after beep.
The PCMU part is happening because even though I am using latest master, my conf is old and I don't have OPUS in my global_codec_prefs var.
However, the original problem of noise after beep in the conference happens with the latest conf using OPUS as well. Easily reproducible on vanilla config on Windows 7. Of course switching to cdquality fixed it as did switching to 12000 in the default conference profile.
Not sure I understand the remark about the ability to hear and understand everyone in the conference. I have three types of clients who all mix and match in the conference - a PSTN termination provider (with SIP connection/PCMU), SIP softphone (PCMU) and Chrome WebRTC (JsSIP). Since the non-WebRTC end points don't support OPUS, I am not sure using it will help in this case. Will it?
Also, if I do use OPUS, any advice on how I can limit the bandwidth it uses? I saw opus.conf.xml but it was not clear how to set a bandwidth limit in it. My users are all in controlled bandwidth environments, so I need to make sure a single user does not use more than 80-100 kpbs (in each direction) even if more is available at the time.
On Mon, Jun 16, 2014 at 8:14 PM, Brian West <brian@freeswitch.org> wrote:
Why aren't you doing OPUS? Are you not liking the ability to hear and understand EVERYONE in the conference? Makes little sense to me, Sounds like your browser has a bug when PCMU is used at 8k I would really love to know what the problem is and the scenario that triggers it. Its unacceptable to say 'Oh set it to 12000 it fixes it'. The reason you have nobody complaining is they've never really experienced WebRTC properly.
You should also be using 1.0.1h, NOT g.
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On Jun 16, 2014, at 9:55 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
My negotiations say I am using PCMU, so since 12000 works for me and 48000 will take extra CPU, any reason I can't just use that?
My system is in production already and nobody's been complaining about voice quality yet.
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