Reply To: RE: sip session timeout, dropped calls

VitalPBX Community Support General Discussion sip session timeout, dropped calls Reply To: RE: sip session timeout, dropped calls

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     ————————– SIP Session-Timers (RFC 4028)————————————
    SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
    This mechanism can detect and reclaim SIP channels that do not terminate through normal
    signaling procedures. Session-Timers can be configured globally or at a user/peer level.
    The operation of Session-Timers is driven by the following configuration parameters:

    • session-timers – Session-Timers feature operates in the following three modes:
      • originate: Request and run session-timers always
      • accept: Run session-timers only when requested by other UA
      • refuse: Do not run session timers in any case

    The default mode of operation is ‘accept’.

    • session-expires – Maximum session refresh interval in seconds. Defaults to 1800 secs.
    • session-minse – Minimum session refresh interval in seconds. Defualts to 90 secs.
    • session-refresher – The session refresher (uac|uas). Defaults to ‘uas‘.
      • uac – Default to the caller initially refreshing when possible
      • uas – Default to the callee initially refreshing when possible

    Note that, due to recommendations in RFC 4028, Asterisk will always honor the other
    endpoint’s preference for who will handle refreshes. Asterisk will never override the
    preferences of the other endpoint. Doing so could result in Asterisk and the endpoint
    fighting over who sends the refreshes. This holds true for the initiation of session
    timers and subsequent re-INVITE requests whether Asterisk is the caller or callee, or
    whether Asterisk is currently the refresher or not.

    Configurations Example:

    session-timers=originate
    session-expires=600
    session-minse=90
    session-refresher=uac

    You may put this kind of configurations in the custom TAB on the SIP Settings module

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