<div dir="ltr"><div><div><div>SIP packet loss is one thing, RTP packet loss is another one.<br></div>One does not necessarily imply the other though, of course, both may happen for a common reason.<br><br></div>What about audio codecs ?<br></div><div>Is it possible to configure things so that you only have a single codec enabled all over your system (trunks, phones, ...) ?<br></div><div>Do you still have audio issues with a single codec ?<br></div></div><div class="gmail_extra"><br><div class="gmail_quote">2017-01-30 17:55 GMT+01:00 Motty Cruz <span dir="ltr"><<a href="mailto:motty.cruz@gmail.com" target="_blank">motty.cruz@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div link="blue" vlink="purple" lang="EN-US"><div class="m_-157384572929130871WordSection1"><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from here: <a href="http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz" target="_blank">http://downloads.asterisk.org/<wbr>pub/telephony/asterisk/<wbr>asterisk-13-current.tar.gz</a> <wbr> <u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">I continue to see errors like this: <u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">[2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission <a href="mailto:56849706-ba96a6d9-817305d0@192.168.125.173" target="_blank">56849706-ba96a6d9-817305d0@<wbr>192.168.125.173</a> for seqno 109 (Critical Request) -- See <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" target="_blank">https://wiki.asterisk.org/<wbr>wiki/display/AST/SIP+<wbr>Retransmissions</a><u></u><u></u></span></p><span class=""><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Packet timed out after 32000ms with no response<u></u><u></u></span></p></span><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">[2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission <a href="mailto:6e3dd238-911e2ac3-f1260152@192.168.125.152" target="_blank">6e3dd238-911e2ac3-f1260152@<wbr>192.168.125.152</a> for seqno 103 (Critical Request) -- See <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" target="_blank">https://wiki.asterisk.org/<wbr>wiki/display/AST/SIP+<wbr>Retransmissions</a><u></u><u></u></span></p><span class=""><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Packet timed out after 32000ms with no response<u></u><u></u></span></p></span><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">[2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission <a href="mailto:ed38f9c8-295a9db-c23f5242@192.168.125.144" target="_blank">ed38f9c8-295a9db-c23f5242@192.<wbr>168.125.144</a> for seqno 103 (Critical Request) -- See <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" target="_blank">https://wiki.asterisk.org/<wbr>wiki/display/AST/SIP+<wbr>Retransmissions</a><u></u><u></u></span></p><span class=""><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Packet timed out after 32000ms with no response<u></u><u></u></span></p></span><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">[2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission <a href="mailto:ef497d11-a81b1c00-8bfbd3bf@192.168.1.244" target="_blank">ef497d11-a81b1c00-8bfbd3bf@<wbr>192.168.1.244</a> for seqno 103 (Critical Request) -- See <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" target="_blank">https://wiki.asterisk.org/<wbr>wiki/display/AST/SIP+<wbr>Retransmissions</a><b><u></u><u></u></b></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Before upgrading to this new server, Asterisk version 1.8 on CentOS 5.9 hardware on both servers were similar in CPU, Memory <u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Any support on this matter is appreciated!<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Thanks, <br>Motty <u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><div style="border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0in 0in 0in"><p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.<wbr>digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-<wbr>bounces@lists.digium.com</a>] <b>On Behalf Of </b>kambiz sharifi<br><b>Sent:</b> Saturday, January 28, 2017 5:13 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Asterisk 13.13.1<u></u><u></u></span></p></div><div><div class="h5"><p class="MsoNormal"><u></u> <u></u></p><div><p class="MsoNormal"><u></u> <u></u></p><div><div><p class="MsoNormal">On Wed, Jan 25, 2017 at 16:00 Olivier <<a href="mailto:oza.4h07@gmail.com" target="_blank">oza.4h07@gmail.com</a>> wrote:<u></u><u></u></p></div><blockquote style="border:none;border-left:solid #cccccc 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in"><div><p class="MsoNormal">What did you exactly upgade ? Asterisk only ? Asterisk and OS ?<br>How did you installed Asterisk 1.8 and 13 ? From source or from package ?<br><br>I would be curious to see what would happen after downgrading back to 1.8.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p><div><p class="MsoNormal">2017-01-24 21:03 GMT+01:00 Motty Cruz <span class="m_-157384572929130871gmailmsg"><<a href="mailto:motty.cruz@gmail.com" target="_blank">motty.cruz@gmail.com</a>></span>:<u></u><u></u></p><div><div><p class="MsoNormal">Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are starting to complaint about packets loss, conversations are choppy! <u></u><u></u></p><p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">PkEI don’t even know where to start looking! Choppy conversations happened within users. I am using sip.conf<u></u><u></u></p><p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">[1091]<u></u><u></u></p><p class="MsoNormal">type=friend<u></u><u></u></p><p class="MsoNormal">context=sip-phone<u></u><u></u></p><p class="MsoNormal">call-limit=2<u></u><u></u></p><p class="MsoNormal">trustrpid=no<u></u><u></u></p><p class="MsoNormal">callerid="dev1" <1091><u></u><u></u></p><p class="MsoNormal">disallow=all<u></u><u></u></p><p class="MsoNormal">allow=ulaw<u></u><u></u></p><p class="MsoNormal">allow=alaw<u></u><u></u></p><p class="MsoNormal">username=1091<u></u><u></u></p><p class="MsoNormal">secret=XXXXX<u></u><u></u></p><p class="MsoNormal">dtmfmode=rfc2833<u></u><u></u></p><p class="MsoNormal">host=dynamic<u></u><u></u></p><p class="MsoNormal">mailbox=10091@default<u></u><u></u></p><p class="MsoNormal">nat=force_rport,comedia<u></u><u></u></p><p class="MsoNormal">canreinvite=no<u></u><u></u></p><p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">extensions.conf<u></u><u></u></p><p class="MsoNormal">exten => 1091,hint,SIP/${EXTEN}<u></u><u></u></p><p class="MsoNormal">exten => 1091,1,Dial(SIP/${EXTEN},15,t)<u></u><u></u></p><p class="MsoNormal">exten => 1091,2,Voicemail(${EXTEN}@<wbr>default,u)<u></u><u></u></p><p class="MsoNormal">exten => 1091,102,Voicemail(${EXTEN}@<wbr>default,b)<u></u><u></u></p><p class="MsoNormal">exten => 1091,103,Hangup<u></u><u></u></p><p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">[2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: <u></u><u></u></p><p class="MsoNormal">Retransmission timeout reached on transmission <a href="mailto:7c803889-63e1b3fe-c2b5ef77@192.168.0.191" target="_blank">7c803889-63e1b3fe-c2b5ef77@<wbr>192.168.0.191</a> for seqno 156 (Critical Request) -- See <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" target="_blank">https://wiki.asterisk.org/<wbr>wiki/display/AST/SIP+<wbr>Retransmissions</a><u></u><u></u></p><p class="MsoNormal">Packet timed out after 32000ms with no response<u></u><u></u></p><p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">any ideas? <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"><span style="color:#888888">Motty<u></u><u></u></span></p></div></div><p class="MsoNormal"><br>--<br><br><br>______________________________<wbr>______________________________<wbr>_________<br><br><br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br><br><br><br><br><br>Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" target="_blank">https://community.asterisk.<wbr>org/</a><br><br><br><br><br><br>New to Asterisk? Start here:<br><br><br> <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" target="_blank">https://wiki.asterisk.org/<wbr>wiki/display/AST/Getting+<wbr>Started</a><br><br><br><br><br><br>asterisk-users mailing list<br><br><br>To UNSUBSCRIBE or update options visit:<br><br><br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/<wbr>mailman/listinfo/asterisk-<wbr>users</a><u></u><u></u></p></div><p class="MsoNormal"><u></u> <u></u></p></div><p class="MsoNormal"><br><br>--<br><br>______________________________<wbr>______________________________<wbr>_________<br><br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br><br><br><br>Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" target="_blank">https://community.asterisk.<wbr>org/</a><br><br><br><br>New to Asterisk? Start here:<br><br> <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" target="_blank">https://wiki.asterisk.org/<wbr>wiki/display/AST/Getting+<wbr>Started</a><br><br><br><br>asterisk-users mailing list<br><br>To UNSUBSCRIBE or update options visit:<br><br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/<wbr>mailman/listinfo/asterisk-<wbr>users</a><u></u><u></u></p></blockquote></div></div></div></div></div></div><br>--<br>
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Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" rel="noreferrer" target="_blank">https://community.asterisk.<wbr>org/</a><br>
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New to Asterisk? Start here:<br>
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