<div dir="ltr">So it sounds like you're writing your own module, How are you doing this hold operation?<div><br></div><div>/b</div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Nov 3, 2016 at 10:53 AM, Miroslav Levanic <span dir="ltr"><<a href="mailto:Miroslav.Levanic@enghouse.com" target="_blank">Miroslav.Levanic@enghouse.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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<p class="MsoNormal">Hello,<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">I’m trying to play from file when other party is set on–hold.
<u></u><u></u></p>
<p class="MsoNormal">reINVITE sent by FreeSwitch contains sendonly attribute which opens possibility that FreeSwitch can play something to other end although other party is on-hold.
<u></u><u></u></p>
<p class="MsoNormal">I’ve used modified switch_ivr_play_file() function which by default prevents sending rtp packets to other side when call is on-hold.<u></u><u></u></p>
<p class="MsoNormal">I’ve encountered rtp stream issue causing distorted audio due to dropped packets on the caller end. <u></u><u></u></p>
<p class="MsoNormal">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.<u></u><u></u></p>
<p class="MsoNormal">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.<u></u><u></u></p>
<p class="MsoNormal">Setting "auto-jitterbuffer-msec" parameter in sofia did not help.<u></u><u></u></p>
<p class="MsoNormal">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. <u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">Thanks,<u></u><u></u></p>
<p class="MsoNormal">Miro<u></u><u></u></p>
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<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr">
<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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