<div dir="ltr">You have ptime 40, does it work the common way (with ptime 20) ? Change it in the client, your client makes an offer with ptime 40. <div>It should not be a problem (40 ms is ok too) , but maybe we get a hint on this one way audio issue. </div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Thu, Jul 9, 2020 at 9:29 PM Paul Mateer <<a href="mailto:Paul.Mateer@outlook.com">Paul.Mateer@outlook.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
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<div>
<p class="MsoNormal"><span lang="EN-US">Don't know if it's the same problem but I incorporated FreeSWITCH code into a product a couple of years ago (master code was marked as 1.9.x) and whilst the (FreeSWITCH) server worked fine, a client built with that version
of the software would only give audio in one direction.<u></u><u></u></span></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">I figured out a fix (to switch_ivr_originate.c) and raised a Jira but didn't provide the fix as I wasn’t sure the change I had made was the best way to resolve the issue. I assumed that someone with a better understanding of the code would
be better placed to determine the most appropriate fix.</p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">Interestingly if I used a much older version of the FreeSWITCH software in the client (like that shipping with the FSClient software) the problem did not occur.</p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">Sent from my Windows 10 device</p>
<p class="MsoNormal"><u></u> <u></u></p>
<div style="border-right:none;border-bottom:none;border-left:none;border-top:1pt solid rgb(225,225,225);padding:3pt 0cm 0cm">
<p class="MsoNormal" style="border:none;padding:0cm"><b>From: </b><a href="mailto:nbhatti@gmail.com" target="_blank">Muhammad Naseer Bhatti</a><br>
<b>Sent: </b>09 July 2020 18:44<br>
<b>To: </b><a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
<b>Subject: </b>[Freeswitch-users] Oneway Audio, can’t send RTP back to UAC</p>
</div>
<p class="MsoNormal"><u></u> <u></u></p>
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<div>
<div style="font-family:Helvetica,Arial;font-size:13px"><br>
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<div style="font-family:Helvetica,Arial;font-size:13px">Hi,</div>
<div style="font-family:Helvetica,Arial;font-size:13px">I can’t seem to be able to figure out why I can’t send RTP to the other side (UAC) of the switch (One way audio?) . I have FreeSWITCH Version 1.10.4-dev git 00113c4 with vanilla config (for testing) on
Public IP address and the other switch is also on Public IP on same subnet. I receive call from SippySoft (UAC) and playing delay_echo application. The call flow is</div>
<div style="font-family:Helvetica,Arial;font-size:13px"><br>
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<div style="font-family:Helvetica,Arial;font-size:13px">SIP UA (NATted) -> SippySwitch (Public IP) - FreSWITCH (Public IP)</div>
<div style="font-family:Helvetica,Arial;font-size:13px"><br>
</div>
<div style="font-family:Helvetica,Arial;font-size:13px">FreeSWITCH after answering the call says </div>
<div style="font-family:Helvetica,Arial;font-size:13px"><br>
</div>
<div style="font-family:Helvetica,Arial;font-size:13px">
<div style="margin:0px">
<div style="margin:0px">2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1]</div>
<div style="margin:0px">2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1]</div>
<div style="margin:0px">2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMU:0:8000:20:64000:1]</div>
<div style="margin:0px">2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5630 Audio Codec Compare [PCMU:0:8000:20:64000:1] is saved as a near-match</div>
<div style="margin:0px">2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5701 Substituting codec PCMU@40i@8000h@1c</div>
<div style="margin:0px">2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:3839 Set Codec sofia/internal/<a href="mailto:04232152273@43.225.99.130" target="_blank">04232152273@43.225.99.130</a> PCMU/8000 40 ms 320 samples 64000 bits 1 channels</div>
<div style="margin:0px"><b><u>2020-07-09 21:48:37.902333 [DEBUG] switch_core_codec.c:111 sofia/internal/<a href="mailto:04232152273@43.225.99.130" target="_blank">04232152273@43.225.99.130</a> Original read codec set to PCMU:0</u></b></div>
<div style="margin:0px"><br>
</div>
<div style="margin:0px">and then starts delay_echo() app. Isn’t Original read codec set to PCMU:0 is a bad thing? Perhaps the reason not able to send RTP back? </div>
<div style="margin:0px"><br>
</div>
<div style="margin:0px">On the other hand, I installed Asterisk 13.34.0, on the same machine, just to prove the point if there is network issue but things work just fine with Asterisk default config. Seem like there is either something not configured (default)
in FreeSWITCH and I can’t seem to be able to find either. SDP in Asterisk and FreeSWITCH both seems to be the same.</div>
<div style="margin:0px"><br>
</div>
<div style="margin:0px">FreeSWITCH SIP Trace is here <a href="https://pastebin.freeswitch.org/view/0925119c" target="_blank">
https://pastebin.freeswitch.org/view/0925119c</a> and console log is here <a href="https://pastebin.freeswitch.org/view/69b5f68c" target="_blank">https://pastebin.freeswitch.org/view/69b5f68c</a></div>
<div style="margin:0px">Asterisk SIP Trace is here <a href="https://pastebin.freeswitch.org/view/f38abc97" target="_blank">https://pastebin.freeswitch.org/view/f38abc97</a> </div>
</div>
</div>
<div></div>
<div><br>
</div>
<div>Appreciate some input to figure out this problem.</div>
<div><br>
</div>
<div><br>
</div>
<div>Thanks,</div>
<div>Naseer</div>
</div>
</div>
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