Luca Bertoncello
2020-Jun-22 15:18 UTC
[asterisk-users] Voice broken during calls (again...)
Am 22.06.2020 um 17:01 schrieb Telium Technical Support:> I don't know if there was a prior email with more details, but.... > > Latency is as important as speed. Have you checked latency between your device and pop? What about QoS at your location, and does your ITSP support/respect QoS?That's a very good idea... Could you suggest me how can I check it? The Gateway is a Linux with Debian 9.> Could problem be inside your network? Have you tested/optimized internal?Really difficult to believe... If I call another VoIP-phone in my network (using the "internal number") the quality is excellent. If I call my wife using the "external number", the quality is very bad... Thanks Luca Bertoncello (lucabert at lucabert.de)
Hello,
try pinging your sip peer ip address following way:
ping -n -M do -s 1300 -i 0.1 -c 100 ${ipaddress}
Post several lines and the statistics.
Were you also thinking about MTU problems? Not very probable, but one
never knows.
Marek
2020-06-22 17:18 GMT+02:00, Luca Bertoncello <lucabert at
lucabert.de>:> Am 22.06.2020 um 17:01 schrieb Telium Technical Support:
>> I don't know if there was a prior email with more details, but....
>>
>> Latency is as important as speed. Have you checked latency between
your
>> device and pop? What about QoS at your location, and does your ITSP
>> support/respect QoS?
>
> That's a very good idea...
> Could you suggest me how can I check it?
> The Gateway is a Linux with Debian 9.
>
>> Could problem be inside your network? Have you tested/optimized
internal?
>
> Really difficult to believe... If I call another VoIP-phone in my
> network (using the "internal number") the quality is excellent.
>
> If I call my wife using the "external number", the quality is
very bad...
>
> 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
Michael Keuter
2020-Jun-22 15:47 UTC
[asterisk-users] Voice broken during calls (again...)
You could also use the 'mtr' command under Linux.> Am 22.06.2020 um 17:41 schrieb Marek Greško <mgresko8 at gmail.com>: > > Hello, > > try pinging your sip peer ip address following way: > > ping -n -M do -s 1300 -i 0.1 -c 100 ${ipaddress} > > Post several lines and the statistics. > > Were you also thinking about MTU problems? Not very probable, but one > never knows. > > Marek > > > 2020-06-22 17:18 GMT+02:00, Luca Bertoncello <lucabert at lucabert.de>: >> Am 22.06.2020 um 17:01 schrieb Telium Technical Support: >>> I don't know if there was a prior email with more details, but.... >>> >>> Latency is as important as speed. Have you checked latency between your >>> device and pop? What about QoS at your location, and does your ITSP >>> support/respect QoS? >> >> That's a very good idea... >> Could you suggest me how can I check it? >> The Gateway is a Linux with Debian 9. >> >>> Could problem be inside your network? Have you tested/optimized internal? >> >> Really difficult to believe... If I call another VoIP-phone in my >> network (using the "internal number") the quality is excellent. >> >> If I call my wife using the "external number", the quality is very bad... >> >> Thanks >> Luca Bertoncello >> (lucabert at lucabert.de)Michael http://www.mksolutions.info
Telium Technical Support
2020-Jun-22 15:51 UTC
[asterisk-users] Voice broken during calls (again...)
Still lots of detail missing, but....likely causes include: 1. Egress latency (does your router/firewall support QoS, are you leaving headroom ) 2. Ingress latency - does your ITSP support it 3. Router/firewall latency - can it keep up with the traffic and packet size. Do you have way too many iptables rules in your Debian box? Between ping and traceroute you can probably get some basic stats. Some speed test websites even report latency, other sites will should tracert/ping from outside in to you. How about putting a phone on the DSL/cable modem directly and calling out...same problem? -----Original Message----- From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Luca Bertoncello Sent: Monday, June 22, 2020 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Voice broken during calls (again...) Am 22.06.2020 um 17:01 schrieb Telium Technical Support:> I don't know if there was a prior email with more details, but.... > > Latency is as important as speed. Have you checked latency between your device and pop? What about QoS at your location, and does your ITSP support/respect QoS?That's a very good idea... Could you suggest me how can I check it? The Gateway is a Linux with Debian 9.> Could problem be inside your network? Have you tested/optimized internal?Really difficult to believe... If I call another VoIP-phone in my network (using the "internal number") the quality is excellent. If I call my wife using the "external number", the quality is very bad... 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
Luca Bertoncello
2020-Jun-22 18:09 UTC
[asterisk-users] Voice broken during calls (again...)
Am 22.06.2020 um 17:41 schrieb Marek Greško: Hi> try pinging your sip peer ip address following way: > > ping -n -M do -s 1300 -i 0.1 -c 100 ${ipaddress} > > Post several lines and the statistics.root at bpi:/etc/asterisk# ping -n -M do -s 1300 -i 0.1 -c 100 tel.t-online.de PING tel.t-online.de (217.0.128.133) 1300(1328) bytes of data. 1308 bytes from 217.0.128.133: icmp_seq=1 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=2 ttl=57 time=17.9 ms 1308 bytes from 217.0.128.133: icmp_seq=3 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=4 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=5 ttl=57 time=18.1 ms 1308 bytes from 217.0.128.133: icmp_seq=6 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=7 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=8 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=9 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=10 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=11 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=12 ttl=57 time=17.7 ms 1308 bytes from 217.0.128.133: icmp_seq=13 ttl=57 time=17.8 ms 1308 bytes from 217.0.128.133: icmp_seq=14 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=15 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=16 ttl=57 time=17.9 ms 1308 bytes from 217.0.128.133: icmp_seq=17 ttl=57 time=18.2 ms 1308 bytes from 217.0.128.133: icmp_seq=18 ttl=57 time=17.9 ms 1308 bytes from 217.0.128.133: icmp_seq=19 ttl=57 time=18.4 ms 1308 bytes from 217.0.128.133: icmp_seq=20 ttl=57 time=17.9 ms 1308 bytes from 217.0.128.133: icmp_seq=21 ttl=57 time=18.2 ms 1308 bytes from 217.0.128.133: icmp_seq=22 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=23 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=24 ttl=57 time=17.8 ms 1308 bytes from 217.0.128.133: icmp_seq=25 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=26 ttl=57 time=18.1 ms 1308 bytes from 217.0.128.133: icmp_seq=27 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=28 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=29 ttl=57 time=18.1 ms 1308 bytes from 217.0.128.133: icmp_seq=30 ttl=57 time=17.9 ms 1308 bytes from 217.0.128.133: icmp_seq=31 ttl=57 time=18.3 ms 1308 bytes from 217.0.128.133: icmp_seq=32 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=33 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=34 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=35 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=36 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=37 ttl=57 time=18.1 ms 1308 bytes from 217.0.128.133: icmp_seq=38 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=39 ttl=57 time=18.0 ms 1308 bytes from 217.0.128.133: icmp_seq=40 ttl=57 time=17.9 ms 1308 bytes from 217.0.128.133: icmp_seq=41 ttl=57 time=17.9 ms 1308 bytes from 217.0.128.133: icmp_seq=42 ttl=57 time=18.1 ms 1308 bytes from 217.0.128.133: icmp_seq=43 ttl=57 time=18.1 ms 1308 bytes from 217.0.128.133: icmp_seq=44 ttl=57 time=18.1 ms 1308 bytes from 217.0.128.133: icmp_seq=45 ttl=57 time=17.9 ms 1308 bytes from 217.0.128.133: icmp_seq=46 ttl=57 time=18.1 ms ^C --- tel.t-online.de ping statistics --- 46 packets transmitted, 46 received, 0% packet loss, time 4527ms rtt min/avg/max/mdev = 17.784/18.058/18.454/0.190 ms, pipe 2 But now I made a test with a friend of mine, and I think the results are very interesting... So, we configured his mobile phone (Android) to use my Asterisk as peer. We created also a VoIP account on the phone. The phone was *NOT* in my WLAN. The friend called my phone (Thomson ST2022 in local LAN). This was a VoIP call inside Asterisk (two peers speaking together). Deutsche Telekom was *NOT* used now! I can hear very good the friend, without "broken voice", but *he* just hear "like a robot with sore throat" and can't understand any word... The same if I call ihm from my phone (via VoIP). I tried to call my wife's phone from my phone (both in the LAN, both Thomson ST2022). Excellent quality in both direction. Last test: I configured my Android phone and added a VoIP-account on my Asterisk, so now I have my Android as peer in my Asterisk. Then I called my friend's phone (also logged in my Asterisk). First test was with my mobile phone in my WLAN and his phone via LTE. Terrible quality on his side (he hear me very bad), good quality on my side (I hear ihm good). Second test with *both my phone and my friend's phone* via LTE: excellent quality in both directions. Conclusion (maybe!): it can *not* be a problem in the DSL connection and *maybe* it is not a problem in the communication with the Server of Deutsche Telekom, since I have many problems to communicate between two peers in local Asterisk if one is over LTE and the other in local LAN (but curiously *not* if both peers are in local LAN or both via LTE). Ergo: this *must* be a problem in my Asterisk... So the questions: 1) can someone confirm or contradict my conclusions? 2) assuming are my conclusions correct, can someone suggest me where can I search the problem? Thanks a lot Luca Bertoncello (lucabert at lucabert.de)