sip proxy

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Asterisk Users 2 Comments

hi all
how can i put a sip proxy; how it differs from asterisk pbx; currently
doing some test on asterisk; thru DSL connection the callers can hear only
one way;asterisk pbx is behind NAT; i am in search of a proper VOIP network
;appreciate some clues in this line
thanks

-kbh

2 thoughts on - sip proxy

  • Depending on your OS and which proxy you want, it may be as simple as
    ‘sudo yum -y install opensips’

    I think the major distinction is ‘who handles the signaling and who
    handles the media.’ A SIP proxy usually only handles the signaling. A PBX
    usually handles both. Note that you can combine a SIP proxy and a RTP
    proxy. Then the distinction is based more on the feature set with a PBX
    having more features relating to handsets.

    A common NAT problem. Using ‘wireshark’ to capture packets and analyze
    them may yield some clues. Recent discussions on this list incriminate SIP
    ‘helpers’ in some routers.

    Every one of ‘them’ is different. How big of a system are you designing?
    Astlinux on a Soekris is great for a couple of simultaneous calls.
    Multiple OpenSIPS hosts fronting a farm of Asterisk hosts could probably
    handle a small nation.

  • Greetings,
    If the calls are esteblished,then SIP did its work.
    The system may RTP problems, NAT may or may not be the casue
    for the issue. Defaults RTP ports for asterisk are UDP 10000-20000,
    You may want to look into that direction.

    Guy Gold