Expert Needed - Call deteriorates after 2 mins in

Discussion in 'General' started by daschenbrener, Sep 10, 2010.

  1. daschenbrener

    Joined:
    Aug 11, 2010
    Messages:
    35
    Likes Received:
    0
    Am an able to make calls fine, but after aprox 2 mins into the call the other end indicates the call is choppy and then eventually cannot hear and I have to end the call.
    The call is going over an ITSP.

    And it makes no difference if the call is incoming or I dial out.

    I can hear them perfectly fine and without any break or anything.

    I have got my internet provider to come and diagnose my feed and they indicate it is fine.
    On elastix 2.0
    aterisk 1.6.2.10

    Not too sure where to start on this one. and direction will be appreciated

    Thanks

    David
     
  2. andyshawn

    Joined:
    Apr 3, 2009
    Messages:
    113
    Likes Received:
    0
    These are my thoughts...

    This sounds like an upload/bandwidth problem. Since its the other end that complains that the call starts to deteriorate, then i am guessing that your upload might be limited or your ISP is doing some sort of traffic shaping.

    Does this happen with every call you make to different individuals? If you make a call to a device on the same LAN, does the same thing happen?
     
  3. daschenbrener

    Joined:
    Aug 11, 2010
    Messages:
    35
    Likes Received:
    0
    No it does not happen on the lan side.

    I am on cable with apparently 1mb upload.

    But never thought to ask if they are limiting my upload.
     
  4. andyshawn

    Joined:
    Apr 3, 2009
    Messages:
    113
    Likes Received:
    0
    Its possible that they are dropping your packets. As you know voice traffic is 'very' sensitive to delays and it needs the required bandwidth in order for the packets to arrive in order and ontime. Since you are using cable internet, this is a shared medium and your upload bandwidth of 1mb is not guaranteed.
    Have you tried making calls late at night and early in the morning, where i'm assuming the network traffic is low?
     
  5. daschenbrener

    Joined:
    Aug 11, 2010
    Messages:
    35
    Likes Received:
    0
    It did not seem to make a difference. This is also with 2 different ITSP also.
     
  6. daschenbrener

    Joined:
    Aug 11, 2010
    Messages:
    35
    Likes Received:
    0
    getting off with my ISP, they indicate all is good and signal and up/down speeds are consistent.
    Could it be a deteriorating switch I have in my network?
    It is a netgear gsm 7324 managed layer 3 switch, wide open without any management on.
     
  7. andyshawn

    Joined:
    Apr 3, 2009
    Messages:
    113
    Likes Received:
    0
    I ruled out your network devices as the problem, because you said calls it works good on your LAN. Where is the server located? Is the server overloaded? Do you know how much memory is in the server?
     
  8. daschenbrener

    Joined:
    Aug 11, 2010
    Messages:
    35
    Likes Received:
    0
    3GB of Ram, 2.8ghz HT intel processor, Sata drives. only 7 users on the machine
     
  9. andyshawn

    Joined:
    Apr 3, 2009
    Messages:
    113
    Likes Received:
    0
    Ok then, the possibility of the server being overworked is eliminated then. I can't think of anything else at the moment. Are all other users experiencing the same problem, if yes, then it might be a problem with your router.
     
  10. daschenbrener

    Joined:
    Aug 11, 2010
    Messages:
    35
    Likes Received:
    0
    Yes they are.
    I am using pfsense for my routing. on a p3 800 with 500 mb of ram
     
  11. andyshawn

    Joined:
    Apr 3, 2009
    Messages:
    113
    Likes Received:
    0
    I think thats where the problem is. I assume you are using the firewall feature and your firewall maybe doing some packet filtering/checking which slows up the process. Is it possible for you to use a regular router just to eliminate pfSense as the problem?
     
  12. daschenbrener

    Joined:
    Aug 11, 2010
    Messages:
    35
    Likes Received:
    0
    Question being, what kind that can handle the traffic?
     
  13. andyshawn

    Joined:
    Apr 3, 2009
    Messages:
    113
    Likes Received:
    0
    I'm think any other router, like a linksys wrt54g2 or a dlink router. I just want to change your gateway for a minute and do some testing.
     
  14. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    JM2CWAE

    pfsense is in many ways more competent than a "regular router"

    That daschenbrener is using a cable company, is in itself suspicious, I believe he is in Jamaica, but here in the US, there are many class actions against the "cable companies" for traffic shaping, many here offer a voip service and usually not SIP, it is not in their commercial interest to allow SIP, One of my personal experiences is that Time-Warner will disrupt your rtp traffic for 10 seconds every 5 minutes, they of course deny it.

    you can see the flow of data by issuing

    rtp debug ip <your provider's ip>

    at the asterisk CLI

    and analyze the synchronicity of the traffic flow as the calls degrade.

    (for postpartum forensics, the same data will be timestamped and available in /var/log/asterisk/full)

    dicko
     
  15. daschenbrener

    Joined:
    Aug 11, 2010
    Messages:
    35
    Likes Received:
    0
    I am in regina, sk,canada. My isp is access communcations.
    They claim they are not traffic shaping at all.


    Also when you indicate

    rtp debug ip <your provider's ip>

    do you mean the ip address i am using for my voip server?
     
  16. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    My apologies I copied and pasted the wrong IP address, Regina is much colder than Jamaica.

    but no, the connection is between your external IP and your VSP so that would be the ip of your Voice Service Provider's server.

    They ALL claim they are not traffic shaping, that does not mean the aren't. You will have to prove it to them, they will then of course deny it again.

    Have you set up the QOS on the pfsense correctly?


    good luck

    dicko
     
  17. daschenbrener

    Joined:
    Aug 11, 2010
    Messages:
    35
    Likes Received:
    0
    for whatever reason in my version of asterisk, 1.6.2.10, rtp debug ip does not work?
     
  18. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    I believe it does, you are are probably just doing it wrong.

    if you are confused and have only one phone call at a time

    rtp debug

    should suffice.
     
  19. daschenbrener

    Joined:
    Aug 11, 2010
    Messages:
    35
    Likes Received:
    0
    the rtp set debug on works, the rtp set debug ip, does not, command always fails
    but that is fine, i can track the log on my calls.

    So what is it that I am looking for in the log?
     
  20. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    when a "Sent RTP packet" is not immediately followed by a "Got RTP packet", watch the seq field.

    if that doesn't happen impeccably you are hosed,

    If you are using asterisk 1.6 then

    rtp set debug on

    but

    rtp set debug ip <the ip address>

    should work just fine. replace <the ip address> with the ip address you are interested in e.. 123.123.123.132


    (my advice as of right now is, don't use Elastix 2.0.? yet)
     

Share This Page