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 -- _____________________________________________________________________ -- 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