Hi list! I use Asterisk 13.14.1 from Debian repository on a DSL from Deutsche Telekom. Asterisk works well, but I have really often an high delay (I understand it since the other party speak some seconds before he hears my question and answer) and sometimes I hear an echo. I really don't know what can I check and what can be the problem. The problem exists since a very long time, but in the last months it got worse... Thank you for your help, I can send abstracts of my configuration, if you say me what should I send. Luca Bertoncello (lucabert at lucabert.de)
On Tuesday 11 June 2019 at 20:32:49, Luca Bertoncello wrote:> Hi list! > > I use Asterisk 13.14.1 from Debian repository on a DSL from Deutsche > Telekom. > > Asterisk works well, but I have really often an high delay (I understand > it since the other party speak some seconds before he hears my question > and answer) and sometimes I hear an echo.I think the main question here is: how are you connecting Asterisk to the telephone system? You mention that you're on DSL from Deutsche Telekom, but is the call going over this DSL link to soem SIP provider, who then connects you to the PSTN, or are you connecting Asterisk locally to the phone line via some ATA device? In fact, it's probably worth outlining your hardware arrangement as much as possible: - what sort of telephone are you using - analogue or SIP? - where is your Asterisk server - on your local network, or hosted elsewhere? - how is Asterisk connected to the PSTN? - are the people you're talking to on analogue landline phones, mobiles, or SIP phones? - anything else you can tell us along these lines would probably be helpful. Oh, and what's the *upstream* bandwidth of your Telekom connection? Antony. -- A few words to be cautious of between American and English: - momentarily - suspenders - chips - pants - jelly - pavement - vest - pint (and gallon) - pissed Please reply to the list; please *don't* CC me.
Am 11.06.2019 um 20:42 schrieb Antony Stone: Hi Antony,> I think the main question here is: how are you connecting Asterisk to the > telephone system?Via VoIP...> You mention that you're on DSL from Deutsche Telekom, but is the call going > over this DSL link to soem SIP provider, who then connects you to the PSTN, or > are you connecting Asterisk locally to the phone line via some ATA device?Deutsche Telekom uses since years just VoIP. No ISDN, PSTN, and so on... :( I'm connecting to the VoIP-Server of Deutsche Telekom via DSL (50Mbps down, 10Mbps up). The other party use VoIP, too, since they are in Germany (and Italy) and here there are just VoIP... Sigh! Now I disabled the jitter (jbenable = no), and I called my father in law. He sayd me, the quality is really better, but I hear sometimes little noises... Any other suggestion? Thanks Luca Bertoncello (lucabert at lucabert.de)
On 11.06.19 at 20:32 Luca Bertoncello wrote:> Hi list! > > I use Asterisk 13.14.1 from Debian repository on a DSL from Deutsche > Telekom. > > Asterisk works well, but I have really often an high delay (I understand > it since the other party speak some seconds before he hears my question > and answer) and sometimes I hear an echo.First of all: I'm using Deutsche Telekom, too (with pjsip on CentOS 7) and don't have this problem. Let me sum up at first what I understand at the moment: - Only VoIP - The problem isn't new. - The problem doesn't happen always, but often. - Asterisk uses the internet IP and doesn't do NAT. - You're using chan_sip - not pjsip - DSL-Line: 50/10 MBit My questions to analyze the problem: - What's the real usable DSL sync (can be seen at the modem)? - Are there any (CRC) errors on the DSL side? How many and in which time? - Deutsche Telekom reports the usable bandwidth during pppoe login. In messages, you can see something like SRU=37868#SRD=102957# (it's an example for a 100 MBit line) (grep messages for "SRU=" after a successful pppoe login) It contains the upload and download bandwidth in kbit/s - Did you configure traffic shaping with tc to be sure that voice packages are always sent at first? - Problem can be seen with different callees or just with one? - Are there any callees the problem never occurred? - Is it "just" a delay or is it choppy, too? - You're using Banana PI - which one exactly? RAM? eth interface manufacturer? What about the load (uptime) of the system when the problem occurs? Is it swapping (what says "free")? - What about the temperature of the device if the problem occurs / not occurs? - Is there any other outbound traffic at the same time? Check with the tool bmon at the ppp0 device and take a look at the upstream. One call creates 50 packages/s (pps) on each direction (if there is no other traffic). It shouldn't fluctuate. - Did you set the correct QoS-type for the outgoing sip and rtp packages? In pjsip, the options are: tos=cs3 cos=3 You can check it with wireshark. The DSCP must be expedited forwarding (or the same you can see for incoming voice packages). - asterisk has an own console, that can be reached with asterisk -r as root. At this point, you can get some information about the quality of a running call. For pjsip it's reporting the following e.g.: *CLI> pjsip show channelstats ...........Receive......... .........Transmit.......... BridgeId ChannelId ........ UpTime.. Codec. Count Lost Pct Jitter Count Lost Pct Jitter RTT.... ========================================================================================================== 5d67cd0b x-0000007e 00:00:39 g722 1296 0 0 0.000 1299 0 0 0.000 0.000 5d67cd0b y-0000007f 00:00:39 alaw 1299 0 0 0.000 1296 0 0 0.000 0.000 Instead of "pjsip show channelstats" you have to use something like sip show [press 2 times tab key] to get the possible commands. Each call generates two entries: one for the call from your local phone to asterisk and the other from asterisk to the ISP. Hope this helps to locate the problem. Michael