[Freeswitch-users] rtp-timer-name / timer issues

Tue Feb 6 18:51:53 UTC 2018

Alright, it looks like CNG may be to blame here.

We were using bridge_generate_comfort_noise to deal with some provider
issues and it may be responding poorly to even the slightest latency. If my
understanding is correct, bridge_generate_comfort_noise will generate a
silence packet into the media stream transparently to the endpoints. The
result would be that if the timer hits and no audio is available on a
channel, CNG is immediately generated and placed into the stream, rendering
jitter buffers on either end fairly useless, since they're still receiving
consistent audio streams (just some packets may be silent)

Does this sound reasonable? Short of timing issues it's the only thing I
can think of that would cause the audio differences between the A and B leg
given that a dedicated host didn't solve the problem either.

Thanks in advance,
Colin

On Tue, Feb 6, 2018 at 10:32 AM, Colin Morelli <colin.morelli at gmail.com>
wrote:

> Tested again on a fresh EC2 instances (c5.xlarge) running Debian Jessie
> (Kernel 3.16.0-4-amd64), since I believe that's the current
> recommendation, with packages installed from the Freeswitch mainline
> (version 1.6.20-37-987c9b9~64bit) and vanilla configs. I am still able to
> reproduce issues where one side's audio recording drops packets that are
> present in the other side. Running out of things to look at here, since I
> was able to repro on baremetal as well.
>
> On Tue, Feb 6, 2018 at 9:04 AM, Colin Morelli <colin.morelli at gmail.com>
> wrote:
>
>> Happens on all browsers.
>>
>> Just want to clarify my previous message, though. I had a call bridged A
>> -> B (A is the WebRTC side, B the PSTN). I recorded both legs of the call
>> individually. On the recording for B, B's audio is clear and smooth. On the
>> recording for A, B's audio has dropped packets that correspond with the
>> logs mentioned on FS. Unless I'm misunderstanding something I believe this
>> should eliminate network/WebRTC/clients as being the issue.
>>
>> On Tue, Feb 6, 2018 at 8:45 AM, Geoff Mina <gmina at connectfirst.com>
>> wrote:
>>
>>> What WebRTC client are you using? Does this happen in all browsers or
>>> just one?
>>>
>>>
>>> On Feb 6, 2018, at 5:09 AM, Colin Morelli <colin.morelli at gmail.com>
>>> wrote:
>>>
>>> Appreciate the input, Brian. I’ll definitely try to avoid setting the
>>> timer option.
>>>
>>> In other news. I deployed The exact same FS instance (same docker
>>> container) on baremetal last night and it experiences the same issue. So,
>>> virtualization does not appear to be the problem. I just can’t figure out
>>> what else would cause this. I’m sure it’s something simple.
>>> On Tue, Feb 6, 2018 at 3:49 AM Brian : <brians at iptel.co> wrote:
>>>
>>>> Hi Colin,
>>>>
>>>> Depending on what you are doing with Freeswitch setting the rtp-timer
>>>> to none can produce all sorts of subtle weirdness. I would advise
>>>> against it. 2 things that I remember from our tests with this was lots
>>>> of blocked / hung calls that would build and need to be HUPed and also
>>>> carriers that would send SDP but no RTP when silence was being sent -
>>>> the call wouldn't progress through dialplan - it just got blocked
>>>> waiting on RTP>
>>>>
>>>> B
>>>>
>>>> On Tue, Feb 6, 2018 at 12:56 AM, Colin Morelli <colin.morelli at gmail.com>
>>>> wrote:
>>>> > Hey list,
>>>> >
>>>> > I'm running FS on EC2 (I know, I know). Having some issues with random
>>>> > packet loss, which I believe almost certainly I have narrowed down to
>>>> timer
>>>> > issues and/or network latency/jitter (seems surprising since I'm using
>>>> > c5.xlarge instances).
>>>> >
>>>> > Behavior is that, during a call, brief pauses or notable audio loss
>>>> will
>>>> > occur. This is on high bandwidth links that are otherwise stable.
>>>> Freeswitch
>>>> > logs with max debug spew out "Hot Hit 1" through "Hot Hit 10" and
>>>> eventually
>>>> > "auto-flush catching up 1 packet(s)" in rapid succession (usually
>>>> going
>>>> > through the cycle 4-5 times) before things settle again. Obviously
>>>> that
>>>> > means a minimum of 4-5 audio packets were dropped within the span of a
>>>> > second which results in considerable audio artifacting.
>>>> >
>>>> > Changing rtp-timer-name to none, which I understand to perform
>>>> synchronous
>>>> > reads of RTP audio (as opposed to timer-based async reads) makes the
>>>> audio
>>>> > notably smoother. That said, I'm having a hard time uncovering the
>>>> > consequences of doing this. Obviously I understand that reads will
>>>> block the
>>>> > RTP thread, but I can't seem to understand the potential
>>>> ramifications of
>>>> > this. Could anyone help clarify?
>>>> >
>>>> > My other question is: assuming "timer while hot" indicates what I
>>>> believe it
>>>> > does (that when the timer hit there was >1 packet in the queue to be
>>>> read),
>>>> > couldn't this issue also just be caused by network jitter, and not
>>>> > necessarily just timer inconsistencies?
>>>> >
>>>> > Thanks in advance.
>>>> >
>>>> > Best,
>>>> > Colin
>>>> >
>>>> >
>>>> >
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>>
>
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