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. CountLost Pct Jitter
CountLost Pct Jitter RTT
===
5d67cd0b x-007e 00:00:39 g722 1296 00 0.000 1299
00 0.000 0.000
5d67cd0b y-007f 00:00:39 alaw 1299 00 0.000 1296
00 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
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