SIP softphone - no audio

Discussion in 'VOIP' started by Peter, Jun 30, 2003.

  1. Peter

    Peter Guest

    I'm trying out SIP softphones, X-Lite and SJPhone, and both does not have
    any audio behind NAT. Everything else seems to work, incoming & outgoing
    calls reach SIP client & Asterisk server, but as soon as I should be hearing
    something, nada.

    I've tried changing SIP client description in Asterisk (nat=yes/no),
    changing proxy type in X-Lite, nothing seems to help. What bothers me is
    that incoming calls do get thru NAT (ringing notification), but audio does
    not. Shouldn't Asterisk be sending/receiving audio on the same UDP port as
    where SIP comms? What could be the problem here? Dumb-headed NAT box?

    Outside NAT everything works just fine. I've googled the web & usenet
    without results.

    TIA,
    Peter
    Peter, Jun 30, 2003
    #1
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  2. Peter

    shido Guest

    You need to use nat = 1 instead of nat = yes

    you sip.conf should be similar to this:

    [general]
    port=5060 ; port we use for sip - you can change this if you want
    binadddr= sticktheipyouwantlisteningforsiphere
    context = inbound ; this is where all the sip calls land
    dtmfmode=inband ; to each his own
    allow=ulaw

    [Joe]
    username=Joe
    type=friend
    secret=boink ; you dont need a secret if you havent set a password on the
    sip device
    host = puttheiphere ; or you can put dynamic if the sip devices ip changes
    nat = 1 ; this is where you were in error.....
    disallow=all ; just says we dont the codec set above
    allow=ulaw ; this says ok we want ulaw
    canreinvite = yes ; if this gives you problems change to no....


    Testing with ulaw first is always best, some softphones only come with
    ulaw....


    -Greg




    "Peter" <> wrote in message
    news:bdo6ke$urdm1$...
    > I'm trying out SIP softphones, X-Lite and SJPhone, and both does not have
    > any audio behind NAT. Everything else seems to work, incoming & outgoing
    > calls reach SIP client & Asterisk server, but as soon as I should be

    hearing
    > something, nada.
    >
    > I've tried changing SIP client description in Asterisk (nat=yes/no),
    > changing proxy type in X-Lite, nothing seems to help. What bothers me is
    > that incoming calls do get thru NAT (ringing notification), but audio does
    > not. Shouldn't Asterisk be sending/receiving audio on the same UDP port as
    > where SIP comms? What could be the problem here? Dumb-headed NAT box?
    >
    > Outside NAT everything works just fine. I've googled the web & usenet
    > without results.
    >
    > TIA,
    > Peter
    >
    >
    shido, Jun 30, 2003
    #2
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  3. Peter

    shido Guest

    last resort...

    If you are still having problems , you need to forward a port on your router
    to the ip of the sip device

    so if your sip/softphone is 192.168.0.5

    open up port 16384 and 5060 udp and send them to your phone...


    16385,16386,16387,16388,16389 if you want the other functions......

    -Greg


    "Peter" <> wrote in message
    news:bdo6ke$urdm1$...
    > I'm trying out SIP softphones, X-Lite and SJPhone, and both does not have
    > any audio behind NAT. Everything else seems to work, incoming & outgoing
    > calls reach SIP client & Asterisk server, but as soon as I should be

    hearing
    > something, nada.
    >
    > I've tried changing SIP client description in Asterisk (nat=yes/no),
    > changing proxy type in X-Lite, nothing seems to help. What bothers me is
    > that incoming calls do get thru NAT (ringing notification), but audio does
    > not. Shouldn't Asterisk be sending/receiving audio on the same UDP port as
    > where SIP comms? What could be the problem here? Dumb-headed NAT box?
    >
    > Outside NAT everything works just fine. I've googled the web & usenet
    > without results.
    >
    > TIA,
    > Peter
    >
    >
    shido, Jun 30, 2003
    #3
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