[asterisk-users] Voice broken during calls (again...)
Hi list! So, now I have a business contract and a technician was here to check the DSL... Nothing found, except that for 50Mbps I need now vectoring. Really nice... A couple of years ago I could get 50Mbps without vectoring. Of course, Deutsche Telekom said nothing about this change... Well, I got it working, and now I have 48Mbps down and 10Mbps up. I _REALLY CAN'T_ believe, that this is not enough... The problem with many little disruptions during calls is always here. I tried changing the codecs and changing some settings in the SIP configuration of the peers. No changes... On the Gateway (Banana PI), where the Asterisk server also runs, the load is about 0.50 during calls and it has a Gbps LAN. I can't believe, the problem is here... @all german users using Telekom: how did you configured your Asterisk? @all: thank you for all your suggestion, I really don't know anymore what I can do... Thanks Luca Bertoncello (lucab...@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
Re: [asterisk-users] Voice broken during calls (again...)
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? Could problem be inside your network? Have you tested/optimized internal? -Original Message- From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luca Bertoncello Sent: Monday, June 22, 2020 10:49 AM To: Asterisk Users Subject: [asterisk-users] Voice broken during calls (again...) Hi list! So, now I have a business contract and a technician was here to check the DSL... Nothing found, except that for 50Mbps I need now vectoring. Really nice... A couple of years ago I could get 50Mbps without vectoring. Of course, Deutsche Telekom said nothing about this change... Well, I got it working, and now I have 48Mbps down and 10Mbps up. I _REALLY CAN'T_ believe, that this is not enough... The problem with many little disruptions during calls is always here. I tried changing the codecs and changing some settings in the SIP configuration of the peers. No changes... On the Gateway (Banana PI), where the Asterisk server also runs, the load is about 0.50 during calls and it has a Gbps LAN. I can't believe, the problem is here... @all german users using Telekom: how did you configured your Asterisk? @all: thank you for all your suggestion, I really don't know anymore what I can do... Thanks Luca Bertoncello (lucab...@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 -- _ -- 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
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 (lucab...@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
Re: [asterisk-users] Voice broken during calls (again...)
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 :
> 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
> (lucab...@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
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Re: [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 :
>
> 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 :
>> 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
>> (lucab...@lucabert.de)
Michael
http://www.mksolutions.info
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Re: [asterisk-users] Voice broken during calls (again...)
Still lots of detail missing, butlikely 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-boun...@lists.digium.com] On Behalf Of Luca Bertoncello Sent: Monday, June 22, 2020 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion 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 (lucab...@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 -- _ -- 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
Re: [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@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 que
Re: [asterisk-users] Voice broken during calls (again...)
Am 22.06.20 um 16:48 schrieb Luca Bertoncello: Hi list! So, now I have a business contract and a technician was here to check the DSL... Nothing found, except that for 50Mbps I need now vectoring. Really nice... A couple of years ago I could get 50Mbps without vectoring. Of course, Deutsche Telekom said nothing about this change... Well, I got it working, and now I have 48Mbps down and 10Mbps up. I _REALLY CAN'T_ believe, that this is not enough... This is enough if you're doing it correctly. But that's your job to do it correctly - not Telekom's one. The problem with many little disruptions during calls is always here. Not surprising. That's most probably not a problem of the provider. VoIP of Deutsche Telekom mostly is pretty perfect regarding voice quality and availability. I tried changing the codecs and changing some settings in the SIP configuration of the peers. No changes... Not surprising. Did you check to prevent transcoding? On the Gateway (Banana PI), where the Asterisk server also runs, the load is about 0.50 during calls and it has a Gbps LAN. What's running on this device on parallel? What about other network traffic - not necessarily to the internet interface? I can't believe, the problem is here... That's irrelevant. You have to ensure, that the driver doesn't have any problems. Reducing the queue sizes of the interface may help. @all german users using Telekom: how did you configured your Asterisk? - At first, you have to trace down the problem and analyze those traces when the problem occurred. This could be done with pcapsipdump[1] on both sides (internal and external). Example: pcapsipdump -i ppp0 -p -d /tmp/pcapsipdump & will trace the connection to Telekom. You have to add another process to another device to trace the internal call. Use Wireshark to analyze the dumps. Wireshark understands VoIP. (I assume you are using SIP / RTP on all legs.) Now you can see on which side the problem happens and how it looks like. - Are you using NAT or is asterisk running on the device which runs the ppp-interface? - What's the modem you are using? What about the wiring between APL and modem? Is it done correctly? [2] - Did you configure prioritization for the up-stream regarding RTP and SIP? This is done with the tc tool. - Did you correctly configure tos? For Deutsche Telekom you may use tos=0xb8 (pjsip). You have to verify it with Wireshark with your traces. You have to set it to the same value as the packages which are received from their server. - You have to use the DNS of Deutsche Telekom which they provide during the ppp-login because they usually provide optimal sip servers for you (regarding distance). You're RTT of ping (18 ms) is pretty bad. I'm having here 5 ms to the primary server (Telekom provides 3). See dig +noall +answer _sip._udp.tel.t-online.de SRV e.g. (don't know the hostname for the business infrastructure) Regards, Michael [1] https://sourceforge.net/projects/pcapsipdump/ [2] https://telekomhilft.telekom.de/t5/Telefonie-Internet/Das-richtige-Kabel-zwischen-APL-und-TAE-Dose/ta-p/3499089 -- _ -- 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
Re: [asterisk-users] Voice broken during calls (again...)
Am 22.06.2020 um 21:30 schrieb Michael Maier: > Did you check to prevent transcoding? could you explain what do you mean and how to check it? >> On the Gateway (Banana PI), where the Asterisk server also runs, the >> load is about 0.50 during calls and it has a Gbps LAN. > > What's running on this device on parallel? What about other network > traffic - not necessarily to the internet interface? On the BananaPI? Nothing other PPP, Bind, NTP, Firewall (iptables) and Asterisk. >> I can't believe, the problem is here... > > That's irrelevant. You have to ensure, that the driver doesn't have any > problems. Reducing the queue sizes of the interface may help. I don't understand what you mean... > - Are you using NAT or is asterisk running on the device which runs the > ppp-interface? Asterisk runs on PPP interface > - What's the modem you are using? What about the wiring between APL and > modem? Is it done correctly? [2] Zyxel VMG1312B30A. It works correctly and using the Internet (upload and download) is not a problem > - Did you configure prioritization for the up-stream regarding RTP and > SIP? This is done with the tc tool. Yes > - Did you correctly configure tos? For Deutsche Telekom you may use > tos=0xb8 (pjsip). You have to verify it with Wireshark with your traces. > You have to set it to the same value as the packages which are received > from their server. I use SIP, not PJSIP... Do I have to do that, too? Which value? > - You have to use the DNS of Deutsche Telekom which they provide during > the ppp-login because they usually provide optimal sip servers for you > (regarding distance). You're RTT of ping (18 ms) is pretty bad. I'm > having here 5 ms to the primary server (Telekom provides 3). See > > dig +noall +answer _sip._udp.tel.t-online.de SRV > > e.g. (don't know the hostname for the business infrastructure) I have a forwarding to the DNS servers of Telekom configured in my bind, since the Gateway has to manage the internal domains, too... Regarding the ping time: wich line do you have? I have a DSL 50Mbps. Maybe your times are better due to a faster line? What is your opinion about the tests I did today with the friend and his phone as VoIP-peer? Thanks Luca Bertoncello (lucab...@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
Re: [asterisk-users] Voice broken during calls (again...)
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 (lucab...@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
Re: [asterisk-users] Voice "broken" during calls
Hello Luca, We are still playing with visualization of your data, but I didn't want you to wait any longer for some results. I think I blame both DT and the Pi :) First, a look at the phone side of your Banana Pi. The first thing we noticed is there were a LOT more packets in one direction (north towards DT) than the other (towards the phone): jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ tcpdump -nr testPhone.pcap src 192.168.200.10 | wc -l reading from file testPhone.pcap, link-type EN10MB (Ethernet) 7951 jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ tcpdump -nr testPhone.pcap dst 192.168.200.10 | wc -l reading from file testPhone.pcap, link-type EN10MB (Ethernet) 3981 Note there are almost twice as many packets headed out. Our tool takes a shot at it: jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ wotinder -I testPhone.pcap input: testPhone.pcap 2020/06/16 10:47:16.047401 INVITE 192.168.200.10:25572 (Luca) -> 192.168.200.1:25572 (sip:035014649215)-(81b17560-c0a80101-0-1798,1) 2020/06/16 10:47:16.112866 DUPINVITE 192.168.200.10:25572 (Luca) -> 192.168.200.1:25572 (sip:035014649215)-(81b17560-c0a80101-0-1798,1) 2020/06/16 10:48:43.690647 BYE 192.168.200.1:25572(sip:035014649215) -> 192.168.200.10:25572(Luca) Session: 81b17560-c0a80101-0-1798 RTP 1 -> 10030 Source total pkts: 7899 (avg err 15934.774414) Dest total pkts: 3943 (avg err 8307.511719) The "average error" is the average departure from exactly 50hz, in microseconds. Basically we are wanting to see a packet every 20,000us, and if it arrives early (because the last one was late) or late, then the absolute value of how far off it was is accumulated, and in the end averaged. Its a bit misleading in this case, because there has clearly been packet loss in one direction, and I am still wrapping my head around why the error isn't much higher (some kind of bug in our packet loss penalties). It does show that from the BPi's perspective, the stream from the phone is NOT very steady. The *average* error was almost a full packet length late (16,000us). Now our tool spits out the raw data (time between packets in us) and we can quickly graph it. I lined up the two legs, but of course you are only seeing half of the second one, and it makes an interesting visual: What on earth is causing the very regular spikes? Roughly every second there seems to be a delay introduced, EVEN FROM THE PHONE ON THE LAN! This worries me that we have asked the Pi to do too much. Perhaps capturing the data and writing it while also running asterisk is causing something to back up regularly. We do prefer to do this kind of analysis from a span port on a switch... But that doesn't explain the missing packets from DT. Similar results on that side: jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ tcpdump -nr testDSL.pcap src 91.49.58.181 | wc -l reading from file testDSL.pcap, link-type LINUX_SLL (Linux cooked) 8048 jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ tcpdump -nr testDSL.pcap dst 91.49.58.181 | wc -l reading from file testDSL.pcap, link-type LINUX_SLL (Linux cooked) 4076 I'm making an assumption that 91.49.58.181 is your side of the DSL, and the packets towards you seem to be missing a lot! I can't explain that as a Pi issue *unless* something funny is happening on the kernel handling inbound public traffic. You mention you are traffic shaping - that could easily be causing something like this. Running our tool on that trace: jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ wotinder -I DSL.pcap input: DSL.pcap 2020/06/16 10:47:16.196746 INVITE 91.49.58.181:25572 (00493514977290) -> 217.0.27.53:5060 (sip:035014649215)-(765cb6164b1c122a3b9c8303600ea367,10036) 2020/06/16 10:47:16.296309 DUPINVITE 91.49.58.181:25572 (00493514977290) -> 217.0.27.53:5060 (sip:035014649215)-(765cb6164b1c122a3b9c8303600ea367,10036) 2020/06/16 10:47:16.357971 DUPINVITE 91.49.58.181:25572 (00493514977290) -> 217.0.27.53:5060 (sip:035014649215)-(765cb6164b1c122a3b9c8303600ea367,10036) 2020/06/16 10:47:16.457280 DUPINVITE 91.49.58.181:25572 (00493514977290) -> 217.0.27.53:5060 (sip:035014649215)-(765cb6164b1c122a3b9c8303600ea367,10036) 2020/06/16 10:48:43.680671 BYE 217.0.27.53:5060(sip:035014649215) -> 91.49.58.181:25572(00493514977290) Session: 765cb6164b1c122a3b9c8303600ea367 RTP 10036 -> 6300 Source total pkts: 7898 (avg err 15771.558594) Dest total pkts: 3943 (avg err 6995.069824) The DUPINVITE packets I think are probably negotiations. I didn't dive into a SIP analysis, but it may be good to look at the SDP and confirm codec selection, etc. Funny enough, a visual of THAT side looks exactly the same: This really is telling I think about the Pi's ability to keep up with everything you are
Re: [asterisk-users] Voice broken during calls (again...)
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 ${ip address of the sip server}
You should receive icmp asking for lowering the packet size.
The LTE phones could have lower MTU and thus overcome PMTU problem.
Marek
2020-06-22 21:48 GMT+02:00, Luca Bertoncello :
> A thing I forgot to report...
> My Asterisk listen on an high port (*not* 5060), since I had many
> problems in the past with someone trying to use my Asterisk with brute
> force attack...
>
> I really don't think, this can be the problem, but better to report all...
>
> Regards
> Luca Bertoncello
> (lucab...@lucabert.de)
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
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Re: [asterisk-users] Voice "broken" during calls
Missing packet from DT could be caused by MTU issue. Marek 2020-06-18 5:41 GMT+02:00, Jeff LaCoursiere : > Hello Luca, > > We are still playing with visualization of your data, but I didn't want > you to wait any longer for some results. I think I blame both DT and > the Pi :) > > First, a look at the phone side of your Banana Pi. The first thing we > noticed is there were a LOT more packets in one direction (north towards > DT) than the other (towards the phone): > > jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ tcpdump -nr > testPhone.pcap src 192.168.200.10 | wc -l > reading from file testPhone.pcap, link-type EN10MB (Ethernet) > 7951 > jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ tcpdump -nr > testPhone.pcap dst 192.168.200.10 | wc -l > reading from file testPhone.pcap, link-type EN10MB (Ethernet) > 3981 > > > Note there are almost twice as many packets headed out. Our tool takes > a shot at it: > > jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ wotinder -I > testPhone.pcap > input: testPhone.pcap > 2020/06/16 10:47:16.047401 INVITE 192.168.200.10:25572 (Luca) -> > 192.168.200.1:25572 (sip:035014649215)-(81b17560-c0a80101-0-1798,1) > 2020/06/16 10:47:16.112866 DUPINVITE 192.168.200.10:25572 (Luca) -> > 192.168.200.1:25572 (sip:035014649215)-(81b17560-c0a80101-0-1798,1) > 2020/06/16 10:48:43.690647 BYE 192.168.200.1:25572(sip:035014649215) > -> 192.168.200.10:25572(Luca) > Session: 81b17560-c0a80101-0-1798 > RTP 1 -> 10030 > Source total pkts: 7899 (avg err 15934.774414) > Dest total pkts: 3943 (avg err 8307.511719) > > The "average error" is the average departure from exactly 50hz, in > microseconds. Basically we are wanting to see a packet every 20,000us, > and if it arrives early (because the last one was late) or late, then > the absolute value of how far off it was is accumulated, and in the end > averaged. Its a bit misleading in this case, because there has clearly > been packet loss in one direction, and I am still wrapping my head > around why the error isn't much higher (some kind of bug in our packet > loss penalties). > > It does show that from the BPi's perspective, the stream from the phone > is NOT very steady. The *average* error was almost a full packet length > late (16,000us). Now our tool spits out the raw data (time between > packets in us) and we can quickly graph it. I lined up the two legs, > but of course you are only seeing half of the second one, and it makes > an interesting visual: > > > What on earth is causing the very regular spikes? Roughly every second > there seems to be a delay introduced, EVEN FROM THE PHONE ON THE LAN! > This worries me that we have asked the Pi to do too much. Perhaps > capturing the data and writing it while also running asterisk is causing > something to back up regularly. We do prefer to do this kind of > analysis from a span port on a switch... > > But that doesn't explain the missing packets from DT. > > Similar results on that side: > > jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ tcpdump -nr > testDSL.pcap src 91.49.58.181 | wc -l > reading from file testDSL.pcap, link-type LINUX_SLL (Linux cooked) > 8048 > jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ tcpdump -nr > testDSL.pcap dst 91.49.58.181 | wc -l > reading from file testDSL.pcap, link-type LINUX_SLL (Linux cooked) > 4076 > > > I'm making an assumption that 91.49.58.181 is your side of the DSL, and > the packets towards you seem to be missing a lot! I can't explain that > as a Pi issue *unless* something funny is happening on the kernel > handling inbound public traffic. You mention you are traffic shaping - > that could easily be causing something like this. Running our tool on > that trace: > > jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ wotinder -I > DSL.pcap > input: DSL.pcap > 2020/06/16 10:47:16.196746 INVITE 91.49.58.181:25572 > (00493514977290) -> 217.0.27.53:5060 > (sip:035014649215)-(765cb6164b1c122a3b9c8303600ea367,10036) > 2020/06/16 10:47:16.296309 DUPINVITE 91.49.58.181:25572 > (00493514977290) -> 217.0.27.53:5060 > (sip:035014649215)-(765cb6164b1c122a3b9c8303600ea367,10036) > 2020/06/16 10:47:16.357971 DUPINVITE 91.49.58.181:25572 > (00493514977290) -> 217.0.27.53:5060 > (sip:035014649215)-(765cb6164b1c122a3b9c8303600ea367,10036) > 2020/06/16 10:47:16.457280 DUPINVITE 91.49.58.181:25572 > (00493514977290) -> 217.0.27.53:5060 > (sip:035014649215)-(765cb6164b1c122a3b9c8303600ea367,10036) > 2020/06/16 10:48:43.680671 BYE 217.0.27.53:5060(sip:035014649215) -> > 91.49.58.181:25572(00493514977290) > Session: 765cb6164b1c122a3b9c8303600ea367 > RTP 10036 -> 6300 > Source total pkts: 7898 (avg err 15771.558594) > Dest total pkts: 3943 (avg err 6995.069824) > > > The D
Re: [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=11...@t-online.de
secret= xx
dtmfmode=rfc2833
host=tel.t-online.de
context=luca_incoming
outboundproxy=tel.t-online.de
port=5060
fromuser=035
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@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 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
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
(lucab...@lucabert.de)
--
_
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Re: [asterisk-users] Voice broken during calls (again...)
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 :
> 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=11...@t-online.de
> secret= xx
> dtmfmode=rfc2833
> host=tel.t-online.de
> context=luca_incoming
> outboundproxy=tel.t-online.de
> port=5060
> fromuser=035
> 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@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 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
> 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
> (lucab...@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
--
_
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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
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Re: [asterisk-users] Voice broken during calls (again...)
Am 22.06.2020 um 22:42 schrieb Marek Greško: Hi Marek, > there is no need to change canreinvite for provider configuration. OK, so I leave it... > 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 root@bpi:~# ping -M do -s 1466 tel.t-online.de PING tel.t-online.de (217.0.128.133) 1466(1494) 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 ^C --- tel.t-online.de ping statistics --- 4 packets transmitted, 0 received, +4 errors, 100% packet loss, time 3077ms Do I have a problem? > 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. I don't understand what you mean, could you explain? Thanks Luca Bertoncello (lucab...@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
Re: [asterisk-users] Voice broken during calls (again...)
Am 23.06.2020 07:27, schrieb Luca Bertoncello: I again 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 I checked it, and I see, that the maximum I can use is a paket size of 1464 with all hosts via IPv4. Via IPv6 I can use higher MTU, but I really can't explain why... Can someone explain me what does it mean, if this is a problem for VoIP and how I can solve it? Thanks Luca Bertoncello (lucab...@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
Re: [asterisk-users] Voice broken during calls (again...)
Am 22.06.2020 20:09, schrieb Luca Bertoncello: A couple of other ideas... 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). I think, the problem with bad quality and broken voice just happens if the peers are in different LANs, since if I call my wife's phone (VLAN "phone") using my mobile phone via SIP (in VLAN "intlan") the quality is bad, but if I call her using my phone in VLAN "phone" or if both peers use SIP via LTE the quality is very good... Could you suggest me something to restrict the problem? Currently, I think the problem can be: 1) on Asterisk 2) on my Gateway/Firewall At home I have many VLANs, that normally *not* communicate together (some exceptions are of course implemented). The phones don't reach the Internet via NAT (VLAN "phone" has no routing in Internet). The mobile phones are in VLAN "intlan", with routing in Internet. Any idea? Thanks Luca Bertoncello (lucab...@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
