[Freeswitch-users] Jitter during on-hold

Brian West brian at freeswitch.org
Thu Nov 3 22:05:03 MSK 2016


So it sounds like you're writing your own module, How are you doing this
hold operation?

/b


On Thu, Nov 3, 2016 at 10:53 AM, Miroslav Levanic <
Miroslav.Levanic at enghouse.com> wrote:

> Hello,
>
>
>
> I’m trying to play from file when other party is set on–hold.
>
> reINVITE sent by FreeSwitch contains sendonly attribute which opens
> possibility that FreeSwitch can play something to other end although other
> party is on-hold.
>
> I’ve used modified switch_ivr_play_file() function which by default
> prevents sending rtp packets to other side when call is on-hold.
>
> I’ve encountered rtp stream issue causing distorted audio due to dropped
> packets on the caller end.
>
> Codec used is G.711 mulaw 8kHz and ptime is 20ms accepted on both side,
> FreeSwitch and softphone. Both endpoints are located in the local LAN.
>
> Wireshark rtp analysis shows that when call is not on-hold, packets comes
> in interval of 20ms as expected. But when the call is set on-hold, average
> delta time between packets is 21ms, causing dropping packet every second
> (1ms extra time multiplied by 50 packets per second) with jitter buffer set
> to 50ms in Wireshark.
>
> Setting "auto-jitterbuffer-msec" parameter in sofia did not help.
>
> Is there a serious reason why I cannot use switch_ivr_play_file() during
> on-hold and what could cause jitter only during on-hold period? When call
> is retrieved from hold, jitter disappears and audio is good again.
>
>
>
> Thanks,
>
> Miro
>
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