SIP packet loss is one thing, RTP packet loss is another one. One does not necessarily imply the other though, of course, both may happen for a common reason.
What about audio codecs ? Is it possible to configure things so that you only have a single codec enabled all over your system (trunks, phones, ...) ? Do you still have audio issues with a single codec ? 2017-01-30 17:55 GMT+01:00 Motty Cruz <motty.c...@gmail.com>: > Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from > here: http://downloads.asterisk.org/pub/telephony/asterisk/ > asterisk-13-current.tar.gz > > > > I continue to see errors like this: > > [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: > Retransmission timeout reached on transmission 56849706-ba96a6d9-817305d0@ > 192.168.125.173 for seqno 109 (Critical Request) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > Packet timed out after 32000ms with no response > > [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: > Retransmission timeout reached on transmission 6e3dd238-911e2ac3-f1260152@ > 192.168.125.152 for seqno 103 (Critical Request) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > Packet timed out after 32000ms with no response > > [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: > Retransmission timeout reached on transmission > ed38f9c8-295a9db-c23f5242@192.168.125.144 for seqno 103 (Critical > Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+ > Retransmissions > > Packet timed out after 32000ms with no response > > [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: > Retransmission timeout reached on transmission ef497d11-a81b1c00-8bfbd3bf@ > 192.168.1.244 for seqno 103 (Critical Request) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > > > Before upgrading to this new server, Asterisk version 1.8 on CentOS 5.9 > hardware on both servers were similar in CPU, Memory > > > > Any support on this matter is appreciated! > > > > Thanks, > Motty > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] *On Behalf Of *kambiz sharifi > *Sent:* Saturday, January 28, 2017 5:13 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Asterisk 13.13.1 > > > > > > On Wed, Jan 25, 2017 at 16:00 Olivier <oza.4...@gmail.com> wrote: > > What did you exactly upgade ? Asterisk only ? Asterisk and OS ? > How did you installed Asterisk 1.8 and 13 ? From source or from package ? > > I would be curious to see what would happen after downgrading back to 1.8. > > > > 2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.c...@gmail.com>: > > Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are > starting to complaint about packets loss, conversations are choppy! > > > > > PkEI don’t even know where to start looking! Choppy conversations happened > within users. I am using sip.conf > > > > [1091] > > type=friend > > context=sip-phone > > call-limit=2 > > trustrpid=no > > callerid="dev1" <1091> > > disallow=all > > allow=ulaw > > allow=alaw > > username=1091 > > secret=XXXXX > > dtmfmode=rfc2833 > > host=dynamic > > mailbox=10091@default > > nat=force_rport,comedia > > canreinvite=no > > > > extensions.conf > > exten => 1091,hint,SIP/${EXTEN} > > exten => 1091,1,Dial(SIP/${EXTEN},15,t) > > exten => 1091,2,Voicemail(${EXTEN}@default,u) > > exten => 1091,102,Voicemail(${EXTEN}@default,b) > > exten => 1091,103,Hangup > > > > [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: > > Retransmission timeout reached on transmission 7c803889-63e1b3fe-c2b5ef77@ > 192.168.0.191 for seqno 156 (Critical Request) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > Packet timed out after 32000ms with no response > > > > any ideas? > > > > Thanks! > > Motty > > > -- > > > _____________________________________________________________________ > > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > > > > > New to Asterisk? Start here: > > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > > _____________________________________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > > > New to Asterisk? Start here: > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users