Would you mind repeating the test with canreinvite=no set for all you phones and mobile phones? What is your upload bitrate? Is it guaranteed? I would try also to test the PMTU: Try: ping -M do -s 2000 ${ip address of the sip server} You should receive icmp asking for lowering the packet size. The LTE phones could have lower MTU and thus overcome PMTU problem. Marek 2020-06-22 21:48 GMT+02:00, Luca Bertoncello <lucabert at lucabert.de>:> A thing I forgot to report... > My Asterisk listen on an high port (*not* 5060), since I had many > problems in the past with someone trying to use my Asterisk with brute > force attack... > > I really don't think, this can be the problem, but better to report all... > > Regards > Luca Bertoncello > (lucabert at lucabert.de) > > -- > _____________________________________________________________________ > -- 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
Luca Bertoncello
2020-Jun-22 20:26 UTC
[asterisk-users] Voice broken during calls (again...)
Am 22.06.2020 um 22:12 schrieb Marek Greško: Hi Marek> Would you mind repeating the test with canreinvite=no set for all you > phones and mobile phones?All my peers have already canreinvite=no... I only have canreinvite=yes on the SIP configuration on the Telekom part: [pbxluca] type=peer defaultuser=111111111 at t-online.de secret= xxxxxxxxxx dtmfmode=rfc2833 host=tel.t-online.de context=luca_incoming outboundproxy=tel.t-online.de port=5060 fromuser=03511111111 fromdomain=tel.t-online.de usereqphone=yes canreinvite=yes insecure=port,invite nat=no qualify=yes qualifyfreq=600 disallow=all allow=alaw allow=ulaw Should I change canreinvite=no there?> What is your upload bitrate? Is it guaranteed?Currently 12Mbps. Guaranteed should be about 10Mbps...> I would try also to test the PMTU: > > Try: > > ping -M do -s 2000 ${ip address of the sip server} > > You should receive icmp asking for lowering the packet size.root at bpi:/etc/asterisk# ping -M do -s 2000 tel.t-online.de PING tel.t-online.de (217.0.128.133) 2000(2028) bytes of data. ping: local error: Message too long, mtu=1492 ping: local error: Message too long, mtu=1492 ping: local error: Message too long, mtu=1492 ping: local error: Message too long, mtu=1492 ping: local error: Message too long, mtu=1492 ping: local error: Message too long, mtu=1492 ^C --- tel.t-online.de ping statistics --- 6 packets transmitted, 0 received, +6 errors, 100% packet loss, time 5103ms Mmmm... it seems not good, isn't it? For information, here the output of ifconfig: dsl0: flags=4305<UP,POINTOPOINT,RUNNING,NOARP,MULTICAST> mtu 1492 inet 93.241.x.y netmask 255.255.255.255 destination 62.156.z.k inet6 fe80::9565:3024:4deb:ebc7 prefixlen 10 scopeid 0x20<link> ppp txqueuelen 3 (Point-to-Point Protocol) RX packets 852397 bytes 480197087 (457.9 MiB) RX errors 0 dropped 0 overruns 0 frame 0 TX packets 967912 bytes 170822532 (162.9 MiB) TX errors 0 dropped 0 overruns 0 carrier 0 collisions 0> The LTE phones could have lower MTU and thus overcome PMTU problem.Should I reduce the MTU?!? Maybe I didn't understood what you mean... Thanks Luca Bertoncello (lucabert at lucabert.de)
Hello, there is no need to change canreinvite for provider configuration. Do not change MTU. Probably there will be another problem. I expect packet size 1466 would pass and higher will have the same result. It would be interesting to make the same test from the outside towards your asterisk with size 2 bytes larger the highest you are able to ping. Marek 2020-06-22 22:26 GMT+02:00, Luca Bertoncello <lucabert at lucabert.de>:> Am 22.06.2020 um 22:12 schrieb Marek Greško: > > Hi Marek > >> Would you mind repeating the test with canreinvite=no set for all you >> phones and mobile phones? > > All my peers have already canreinvite=no... > I only have canreinvite=yes on the SIP configuration on the Telekom part: > > [pbxluca] > type=peer > defaultuser=111111111 at t-online.de > secret= xxxxxxxxxx > dtmfmode=rfc2833 > host=tel.t-online.de > context=luca_incoming > outboundproxy=tel.t-online.de > port=5060 > fromuser=03511111111 > fromdomain=tel.t-online.de > usereqphone=yes > canreinvite=yes > insecure=port,invite > nat=no > qualify=yes > qualifyfreq=600 > disallow=all > allow=alaw > allow=ulaw > > Should I change canreinvite=no there? > >> What is your upload bitrate? Is it guaranteed? > > Currently 12Mbps. Guaranteed should be about 10Mbps... > >> I would try also to test the PMTU: >> >> Try: >> >> ping -M do -s 2000 ${ip address of the sip server} >> >> You should receive icmp asking for lowering the packet size. > > root at bpi:/etc/asterisk# ping -M do -s 2000 tel.t-online.de > PING tel.t-online.de (217.0.128.133) 2000(2028) bytes of data. > ping: local error: Message too long, mtu=1492 > ping: local error: Message too long, mtu=1492 > ping: local error: Message too long, mtu=1492 > ping: local error: Message too long, mtu=1492 > ping: local error: Message too long, mtu=1492 > ping: local error: Message too long, mtu=1492 > ^C > --- tel.t-online.de ping statistics --- > 6 packets transmitted, 0 received, +6 errors, 100% packet loss, time 5103ms > > Mmmm... it seems not good, isn't it? > > For information, here the output of ifconfig: > > dsl0: flags=4305<UP,POINTOPOINT,RUNNING,NOARP,MULTICAST> mtu 1492 > inet 93.241.x.y netmask 255.255.255.255 destination 62.156.z.k > inet6 fe80::9565:3024:4deb:ebc7 prefixlen 10 scopeid 0x20<link> > ppp txqueuelen 3 (Point-to-Point Protocol) > RX packets 852397 bytes 480197087 (457.9 MiB) > RX errors 0 dropped 0 overruns 0 frame 0 > TX packets 967912 bytes 170822532 (162.9 MiB) > TX errors 0 dropped 0 overruns 0 carrier 0 collisions 0 > >> The LTE phones could have lower MTU and thus overcome PMTU problem. > > Should I reduce the MTU?!? > Maybe I didn't understood what you mean... > > Thanks > Luca Bertoncello > (lucabert at lucabert.de) > > -- > _____________________________________________________________________ > -- 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