<html><head></head><body>There is No firewall. Already turned off for the sake of testing. on the Same machine just turn off FreeSWITCH and started Asterisk, able to hear audio both ways. Switch back to FreeSWITCH and get one way only. When capturing RTP I can hear far (Play audio stream in Wireshark) but FreeSWITCH won’t try to send any RTPs out. <br> <br><div class="gmail_signature"></div> <div class="gmail_quote" style="color:black"><br>From: <span style="color:black">Dragos Oancea</span> <a href="mailto:dragos@freeswitch.org"><dragos@freeswitch.org></a><br>Reply: <span style="color:black">FreeSWITCH Users Help</span> <a href="mailto:freeswitch-users@lists.freeswitch.org"><freeswitch-users@lists.freeswitch.org></a><br>Date: <span style="color:black">July 9, 2020 at 22:55:22</span><br>To: <span style="color:black">FreeSWITCH Users Help</span> <a href="mailto:freeswitch-users@lists.freeswitch.org"><freeswitch-users@lists.freeswitch.org></a><br>Subject: <span style="color:black"> Re: [Freeswitch-users] Oneway Audio, can’t send RTP back to UAC <br></span></div><br> <blockquote type="cite" class="gmail_quote"><span><div><div></div><div>


<title></title>


<div dir="ltr">PCMU:0 is normal. 0 is the payload type for PCMU.
<div>Perhaps you have the firewall enabled on the machines
themselves ?  Flush your iptables rules, just to see if it
works.<br></div>
<div><br></div>
<div><br></div>
</div>
<br>
<div class="gmail_quote">
<div dir="ltr" class="gmail_attr">On Thu, Jul 9, 2020 at 8:17 PM
Muhammad Naseer Bhatti <<a href="mailto:nbhatti@gmail.com">nbhatti@gmail.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">
<div>
<div style="font-family:Helvetica,Arial;font-size:13px">
<br></div>
<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></div>
<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><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>
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