Am 14.06.2020 um 17:05 schrieb Antony Stone: Hi Antony,> You mean that the Thomson phone is registering to Deutsche Telekom? > > I thought it was registering to your Asterisk server.Sorry, I didn't read correctly your test 2b... Normally my Thomson phone is registering to my Asterisk server. I tried to register the Thomson phone directly to Telekom's server, to check if the problem could be in my Asterisk...> Maybe it would be a good idea to tell *exactly* what your network setup is, > because I'd certainly assumed something that's clearly not true; maybe others > here have as well.Well, I'll try: - DSL-Modem, connected to a BananaPI with Debian 9 - On the BananaPI, PPPoE to connect to the Internet, iptables and some scripts to manage the Gateway and Firewall - Many VLANs, some of them can use the Internet via NAT - The phones are in an own VLAN without any routing to the Internet (exception for my phone was temporarily made to allow the tests) - In the phone's VLAN there is the Asterisk server, running on the same BananaPI the act as Gateway/Firewall - Mobile phone connected via WLAN in the same VLAN used from the PCs, and with routing to the Internet via NAT> Basically, what SIP phones (hardware or software) are you using, what are they > registering to, and what role is Asterisk playing in all of this? How do > calls to/from the public phone network get routed from/to your telephones?We have two phones Thomson ST2022, registering to the Asterisk server. The Asterisk server registers to Deutsche Telekom and MessageNet. All calls are normally routed by Asterisk to Deutsche Telekom. Some *incoming* calls to an italian number arrives via MessageNet and will be directed to my Thomson phone. What I tried connecting the phone directly to the Internet and the servers of Deutsche Telekom was just a test, not the normal situation. Do I have explained my current situation? Of course I can send extract of configurations, if needed... What I'll do tomorrow with a test phone is: 1) connecting it to my Asterisk and try to make a call 2) connecting it directly to the servers of Deutsche Telekom (using my network) and try to make a call Thanks a lot for your help Luca Bertoncello (lucabert at lucabert.de)
Am 14.06.2020 um 17:33 schrieb Luca Bertoncello: Hi So, I got a phone (Elmeg IP290) from a collegue and tested it...> What I'll do tomorrow with a test phone is: > > 1) connecting it to my Asterisk and try to make a call > 2) connecting it directly to the servers of Deutsche Telekom (using my > network) and try to make a callAbsolutly *no changes* on the behaviour compared with my Thomsons... I try to summarize: 1) Phones are not the problem, since 3 phones of 2 different companies/model have the same issue. 2) Asterisk seems not to be the problem, too, since I have the same behaviour if I connect to phone directly to the server of Deutsche Telekom. 3) Traffic shaping seems not to be the problem, too, since I tried to deactivate it. 4) The problem happens *only* on active call, not by voicemail. 4a) To test it I read a text and my partner just listen it, and then he read a text and I listen it. *No* simulaneously speak! 5) A *single call* (since I couldn't reproduce it anymore), made using my Android phone as SIP-client connected to my Asterisk, had not the problem. Any other try to call someone using my mobile phone via SIP had the problem. I could *not* test connecting to the server of Deutsche Telekom using the Internet connection of someone other, since Telekom bounds my VoIP-login to my IP. I really think, the problem should be by Deutsche Telekom... What is your opinion? Do you see some other tests I should try? Thanks a lot Luca Bertoncello (lucabert at lucabert.de)
Hi, We are working on a product to analyze pcap files of VoIP calls. So far it does a reasonable job of analyzing the frequency distribution of packets in both directions, pointing out which direction packet loss / bad jitter occurs. If you can trap the traffic on the outside and the inside of your Banana Pi and send me the pcap files, I would be happy to run it through our analyzer as further information for you. If it shows DTK isn't sending packets when it should, that will be obvious, and you can send to them as solid evidence of their guilt :) Beyond that, are you certain you aren't taxing the Banana Pi? It really *should* be able to handle a single call while handling your LAN's routing/firewall tasks, but you are probably skating the edge. The results of the above might point out that the Pi isn't *sending* packets it should be, or sending them way late, in which case the issue is actually your hardware. Cheers, *Jeff LaCoursiere* STRATUSTALK, INC. / CTO Phone: *+1 703.496.4990 x108* Mobile: *+1 815.546.6599* Email: *jeff at stratustalk.com* <mailto:jeff at stratustalk.com> Website: *https://www.stratustalk.com* Address: *One Freedom Square 13th Floor Reston, VA 20190* <https://www.facebook.com/jeff.lacoursiere> <https://linkedin.com/in/jeff-lacoursiere-884361> <https://www.twitter.com/stratustalk> On 6/15/20 11:55 AM, Luca Bertoncello wrote:> Am 14.06.2020 um 17:33 schrieb Luca Bertoncello: > > Hi > > So, I got a phone (Elmeg IP290) from a collegue and tested it... > >> What I'll do tomorrow with a test phone is: >> >> 1) connecting it to my Asterisk and try to make a call >> 2) connecting it directly to the servers of Deutsche Telekom (using my >> network) and try to make a call > Absolutly *no changes* on the behaviour compared with my Thomsons... > > I try to summarize: > > 1) Phones are not the problem, since 3 phones of 2 different > companies/model have the same issue. > 2) Asterisk seems not to be the problem, too, since I have the same > behaviour if I connect to phone directly to the server of Deutsche Telekom. > 3) Traffic shaping seems not to be the problem, too, since I tried to > deactivate it. > 4) The problem happens *only* on active call, not by voicemail. > 4a) To test it I read a text and my partner just listen it, and then he > read a text and I listen it. *No* simulaneously speak! > 5) A *single call* (since I couldn't reproduce it anymore), made using > my Android phone as SIP-client connected to my Asterisk, had not the > problem. Any other try to call someone using my mobile phone via SIP had > the problem. > > I could *not* test connecting to the server of Deutsche Telekom using > the Internet connection of someone other, since Telekom bounds my > VoIP-login to my IP. > > I really think, the problem should be by Deutsche Telekom... > > What is your opinion? Do you see some other tests I should try? > > Thanks a lot > Luca Bertoncello > (lucabert at lucabert.de) >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200615/0ccecfc9/attachment.html> -------------- next part -------------- A non-text attachment was scrubbed... Name: jeff.vcf Type: text/x-vcard Size: 321 bytes Desc: not available URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200615/0ccecfc9/attachment.vcf>
On Monday 15 June 2020 at 18:55:23, Luca Bertoncello wrote:> Absolutly *no changes* on the behaviour compared with my Thomsons...Okay, I'm glad we can rule out the specific make / model of phone - that would have been bizarre.> I try to summarize: > > 1) Phones are not the problem, since 3 phones of 2 different > companies/model have the same issue.Good (if you see what I mean).> 2) Asterisk seems not to be the problem, too, since I have the same > behaviour if I connect to phone directly to the server of Deutsche Telekom.Is that also via the Banana, or with the phone directly on a DSL modem?> 3) Traffic shaping seems not to be the problem, too, since I tried to > deactivate it.Good test / check.> 4) The problem happens *only* on active call, not by voicemail.So, only when there are two SIP clients active on each side of the Asterisk server...> 4a) To test it I read a text and my partner just listen it, and then he > read a text and I listen it. *No* simulaneously speak!But, what were the results - each of you could hear the other perfectly well? This sounds interesting - more ideas below.> 5) A *single call* (since I couldn't reproduce it anymore), made using > my Android phone as SIP-client connected to my Asterisk, had not the > problem. Any other try to call someone using my mobile phone via SIP had > the problem.You seem to have the problem in general, so a single (or small number of) instances of no problem doesn't mean there isn't something to be resolved.> I could *not* test connecting to the server of Deutsche Telekom using > the Internet connection of someone other, since Telekom bounds my > VoIP-login to my IP.Right.> I really think, the problem should be by Deutsche Telekom...Especially since you say you do not get the problem when you have calls in via Messagenet for your Italian calls.> What is your opinion? Do you see some other tests I should try?Yes. I'm intrigued by the "only one party speaking at a time" test you did. What happens if: a) you call someone external, speak for about 30 seconds without them making any sound, then they start speaking *at the same time as you*, then you stop talking and they carry on. b) exactly the same, except this time they call you, so it's an inbound call. Do you get good quality while only one person speaks, and bad while both do? Does the quality return to good when one person stops speaking? Regards, Antony. -- f u cn rd ths, u cn gt a gd jb n nx prgrmmng Please reply to the list; please *don't* CC me.