Luca Bertoncello
2020-Jun-22 14:48 UTC
[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 (lucabert at lucabert.de)
Telium Technical Support
2020-Jun-22 15:01 UTC
[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-bounces at lists.digium.com] On
Behalf Of Luca Bertoncello
Sent: Monday, June 22, 2020 10:49 AM
To: Asterisk Users <asterisk-users at lists.digium.com>
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
(lucabert at lucabert.de)
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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)
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
Luca Bertoncello
2020-Jun-22 19:44 UTC
[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 (lucabert at lucabert.de)
Luca Bertoncello
2020-Jul-03 17:57 UTC
[asterisk-users] Voice broken during calls (again...)
Hi list! Am 22.06.2020 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... > > 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...So, now I know what was the problem and I solved it... The problem was: the Banana PI... :( I checked it with mtr and I see really bad times to communicate with other devices im same networks (~2 - 380 ms!!). Many tries with other Switch ports and so on didn't solved the problem. So I bought a mini PC and I configured it as Firewall with Asterisk. mtr give now really good times (~0.2 - 0.4 ms). And Asterisk works very good... I tried right now with my father in law and the communication was excellent, without any disruptions. So, I really thank you for the idea, that my Banana PI can be the problem. It was! ;) Have a nice weekend! Luca Bertoncello (lucabert at lucabert.de)
On 03.07.20 at 19:57 Luca Bertoncello wrote: [...]>> 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... > > So, now I know what was the problem and I solved it... > > The problem was: the Banana PI... :(Glad you could find and solve the problem.> I checked it with mtr and I see really bad times to communicate with > other devices im same networks (~2 - 380 ms!!). > Many tries with other Switch ports and so on didn't solved the problem.Yeah, that's what I already thought for myself. VoIP is (based on its realtime nature) extremely picky about network interfaces (or even complete hardware of the system) and their drivers and the corresponding configuration. But most of the people can't / won't believe it. Many of them (NICs) are pretty broken (sometimes the nic hardware, sometimes both hard- and software). Even APU 1 or 2 devices don't work reliably for VoIP with the standard in kernel drivers or with default configuration here. I always had / have to use other drivers / kernels / configurations to get a proper and reliable rtp stream - even over hours. Regards Michael