concurrent channels limit

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Asterisk Users 6 Comments

Hi,
I’ve some problem setting up Asterisk 1.8:
the configuration of the server is
8 core Intel(R) Xeon(R) CPU E5310 @ 1.60GHz
4GB of ram
CentOS release 6.2 (Final)
asterisk-1.8.10.1

I’m using sipp to do some tests, the command I use is “sipp -sf
/root/uac_pcap_G729.xml -s 17000 192.168.200.64 -l 1000″
the file “uac_pcap_G729.xml” just establishes a call, wait 120 seconds
and hangup.

Finally the problem is: I cannot manage more than 80 concurrent calls.
but I know that is a limit too low, the same machine in the production
environment, with the same configurations and asterisk 1.4 manage up to
400 concurrent calls.

How can I find my real upper call limit?

tnx

6 thoughts on - concurrent channels limit

  • Check the sip.conf.sample file. I think it is the call-limit parameter that
    is getting you. The sample file should tell you what the default is.
    Another possibility is that your rtp range is set too low; the “normal”
    range is 10000-20000, which allows for 2500 calls(or 5000 if you set other
    things “correctly”).

  • Asterisk says to process the call correctly:

    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5

  • Ok, this was a stupid thing (my fault), with -r 1000 I get easily 1000
    concurrent calls that terminate in 20 seconds.
    This calls just answer, play a file the first 2 seconds and then wait.
    Then sipp close because of two many errors, this is the log:

    sipp: The following events occured:
    2012-03-30>—–15:17:07:081>—1333117027.081757: Discarding
    message which can’t be mapped to a known SIPp call:
    BYE sip:sipp@192.168.200.185:52281 SIP/2.0^M
    Via: SIP/2.0/UDP 192.168.200.64:5060;branch=z9hG4bK2a711f73;rport^M
    Max-Forwards: 70^M
    Call-ID: 15-2001@192.168.200.185^M
    CSeq: 102 BYE^M
    User-Agent: Asterisk PBX 1.8.11.0^M
    X-Asterisk-HangupCause: Normal Clearing^M
    X-Asterisk-HangupCauseCode: 16^M
    Content-Length: 0^M
    ^M
    .
    2012-03-30>—–15:17:07:580>—1333117027.580847: Discarding
    message which can’t be mapped to a known SIPp call:
    BYE sip:sipp@192.168.200.185:52281 SIP/2.0^M
    Via: SIP/2.0/UDP 192.168.200.64:5060;branch=z9hG4bK2a711f73;rport^M
    Max-Forwards: 70^M
    Call-ID: 15-2001@192.168.200.185^M
    CSeq: 102 BYE^M
    User-Agent: Asterisk PBX 1.8.11.0^M
    X-Asterisk-HangupCause: Normal Clearing^M
    X-Asterisk-HangupCauseCode: 16^M
    Content-Length: 0^M
    ^M
    .
    2012-03-30>—–15:17:07:982>—1333117027.982422: Discarding
    message which can’t be mapped to a known SIPp call:
    BYE sip:sipp@192.168.200.185:38844 SIP/2.0^M
    Via: SIP/2.0/UDP 192.168.200.64:5060;branch=z9hG4bK66a86c70;rport^M
    Max-Forwards: 70^M
    Call-ID: 9-2001@192.168.200.185^M
    CSeq: 102 BYE^M
    User-Agent: Asterisk PBX 1.8.11.0^M
    X-Asterisk-HangupCause: Normal Clearing^M
    X-Asterisk-HangupCauseCode: 16^M
    Content-Length: 0^M
    ^M
    .
    2012-03-30>—–15:17:08:504>—1333117028.504334: Unable to get a
    UDP socket (3).

    But if I change the dialplan, remove background and wait functions, add
    play with a g729 audio file instead, I could do again just 80 concurrent
    call.

  • no, it’s a set of script that I’m supposed to update. However the result
    will be similar.