7 thoughts on - High Delay And Some 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.

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• Am 11.06.2019 um 20:42 schrieb Antony Stone:

  Hi Antony,

  Via VoIP…

  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@lucabert.de)

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• Well, same as Net Cologne here where I am, but the cable modem I have still has PSTN sockets on it so you can connect analogue phones to it as well as speaking SIP to it. I wasn’t sure which you might be doing with your Asterisk.

  So, you have a SIP phone, connected to an Asterisk server on your local network, which then connects to D Telekom’s SIP server over the DSL line?

  Are they also using a SIP phone?

  Do they also have an Asterisk server on their local network?

  Have you considered trying some tool such as http://sipcapture.org/#about to see if you can identify where the latency comes in?

  Antony.

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• Am 11.06.2019 um 21:10 schrieb Antony Stone:

  Hi,

  Correct!

  My mother yes, my father in law uses an ISDN phone connected to a FritzBox that convert the signal in VoIP.

  I must say, that I’m not an expert in VoIP, so I really don’t know this tool and don’t have any idea how to analyze the problem…

  Thanks Luca Bertoncello
  (lucabert@lucabert.de)

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• Well, my starting point, given the hardware setup you’ve confirmed above, would be to plug an analogue phone into your FritzBox (assuming that’s what DT gave you) and see whether the problem exists without Asterisk in the picture at all.

  The second thing I would try is to put the SIP credentials given to you by DT
  into the SIP phone itself (most can support at least two lines, so you don’t need to over-write the credentials for your Asterisk server account) and again, soo whether the problem persists with the Asterisk server removed from the signal path.

  That will at least tell you whether Asterisk is causing the problem, because if it isn’t:

  a) there isn’t much you can do about it except report it to DT, and

  b) there’s very little the good people here on this mailing list will be able to help you with.

  Just out of interest, what hardware are you running Asterisk on? It’s unlikely to be the cause of the problem, because I’ve run it on Raspberry Pies for very small setups such as yours, but it might still be useful to know.

  Regards,

  Antony.

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• Am 11.06.2019 um 21:28 schrieb Antony Stone:

  Hi,

  I don’t have any FritzBox. I have a little BananaPI with Debian 9 configured as Router and connected to a DSL-Modem. On the BananaPI I installed Asterisk, to have it directly connected to the Internet.

  I can try it… Now it’s too late for the test. I’ll try tomorrow.

  Bwahahahahahah!!!! The technician of DT, at least the people answering the Hotline, don’t have any idea _WHAT_ is VoIP and so on… They only can say “you have to power off your FritzBox, wait 30 seconds and power it on again”. If I say, that I don’t have any FritzBox they give a Brain core dumped…

  Really a pity… 🙁

  As I said, I have a BananaPI with a Debian 9, minimal installed from me with some scripts to manage the DSL. Asterisk was installed from Debian Repositories.

  Thanks Luca Bertoncello
  (lucabert@lucabert.de)

  —

• 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

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