Am 13.06.2020 um 22:09 schrieb Antony Stone: Hi Antony
> You are *assuming* that it's the codec causing the difference. Well, I really don't know what I can think, now... > We don't know that. > > Let me get this clear, to make sure I understand (differences emphasised): > > 1. You use *a VoIP softphone app* on your mobile, which is registered by SIP, > to your Asterisk server, over your home *wireless network*, to place a call > to > some external number, you have a conversation and *the quality is excellent*. > > 2. You use your *Thomson ST2022*, which is also registered by SIP, to your > home Asterisk server, over your home *cabled* network, to place a call to > some > (the same???) external number, you have a conversation and the quality is > *not > excellent*. > > > Is that an accurate summary of your situation? Not really... 1) I have an Android phone, using the integrated Android VoIP-subsystem, connected to my Asterisk at home, over LTE or other network *outside my home network*. Today I called my mother using this method (I was in the home network of my parents in law, about 20km von my home network, so definitly *not* in my wireless...). The quality was excellent and it was confirmed from my father in law, too... 2) I have a Thomson ST2022 connected to my Asterisk over Ethernet (cabled network). If I call for example my mother or my parents in law, the conversation is "broken", eg: both partner can hear little "interruption", about 1/10 seconds in the conversation... This is the situation... I tried to connect the Thomson ST2022 directly to the server of Deutsche Telekom via VoIP (excluding the Asterisk, but of couse using NAT, since the phone does not have a public IP but just an IP in my internal network) and then I called my father in law. Same problem... :( I didn't get my Android phone connected to the server of Deutsche Telekom to check how it works *outside my home network*... Not sure why it doesn't work... Some other information: 1) Asterisk runs on a Linux-Box (on a BananaPI) with Debian 10. Asterisk was installed from Debian repositories. 2) The Linux-Box is directly connected to the Internet (no NAT) with a DSL-Modem and PPPoE. Public IPv4 and IPv6 addresses are configured in a network interface of the Linux-Box. 3) I use iptables+tc to manage a traffic shaping, privileging the VoIP connection. If you want, I have no problem to send the traffic-shaping-script to the list. 4) The DSL connection has a speed of 50Mbps down and 10Mbps up, and I really think, it should be enough... 5) The phones are connected with Gbps-Ethernet to the Linux-Box. 6) On my Asterisk I configured a second VoIP-Provider (MessageNet, in Italy), but just to *receive* calls. My contract with MessageNet does not allow me the call someone using this connection. If someone calls my number by MessageNet, I have the same problem I have with Deutsche Telekom, altought not so strong, eg. the "interruptions" are not so frequent as by calls via Deutsche Telekom... Btw: by MessageNet I must use *gsm* as Codec, otherwise a connection will be extablished, but no Voice can be heared... I really appreciate any idea. Of course, it could be possible that there is a problem on Telekom-side, but it does not explain why I have the same problems, altought not often as by Telekom, by MessageNet, too... Thanks a lot 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