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<p class="MsoNormal">Hello,<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I’m trying to play from file when other party is set on–hold.
<o:p></o:p></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.
<o:p></o:p></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.<o:p></o:p></p>
<p class="MsoNormal">I’ve encountered rtp stream issue causing distorted audio due to dropped packets on the caller end. <o:p></o:p></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.<o:p></o:p></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.<o:p></o:p></p>
<p class="MsoNormal">Setting "auto-jitterbuffer-msec" parameter in sofia did not help.<o:p></o:p></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. <o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thanks,<o:p></o:p></p>
<p class="MsoNormal">Miro<o:p></o:p></p>
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