Reproducible ReInvites Sent By UAS After Exactly 900s Despite Session-timers=refuse

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Asterisk Users 3 Comments

Hello!

I’m facing ReInvites as caller from UAS despite configured session-timers=refuse (which can be seen in the SIP trace) always after
900s. (The behavior is the same if session-timers is set to accept).

This just happens with one provider (German Telekom to callee at kabelbw).

– The incoming ReInvite is answered immediately by asterisk (Status 100
/ Status 200 – 0.02s). Media stream is ok.

– 0.04s after sending of Status 200, the media stream from callee is
stopped.

– 0.11s after the Status ok package has been sent, the Ack package of
the UAS can be seen.

– 10s after the arrival of the ack package, UAS sends options packages,
every 10s one package. Each of these packages is immediately answered
by asterisk with Status 200 ok.

– After 31s seconds, asterisk drops the call because of lack of rtp
stream from callee.

Used asterisk version is 13.13.1.

Does anybody has any idea about what’s going on here? This problem isn’t just an actual problem, it can be seen since a few weeks now and is
*always* reproducible (I can provide a trace). I don’t think there is a network problem, because the audio quality before is really good. There isn’t any firewall related problem, too – there are no dropped packages at all.

Thanks, Michael

3 thoughts on - Reproducible ReInvites Sent By UAS After Exactly 900s Despite Session-timers=refuse

  • I tested the exactly same call with another provider (Easybell). Easybell doesn’t support timer at all.

    Via Easybell, the call is dropped after 15 minutes, too, but now, it is ended with RTCP 203 (Good bye) by the callee. 10s later, the callee sends a SIP bye.

  • Another test showed the following behavior:

    – first ReInvite by UAS
    – Trying by UAC
    – OK SDP by UAC (0.02s after ReInvite)
    – ACK by UAS (0.1s after ReInivte)

    – 2 rtp packages arrived from callee, afterwards, there can be seen no
    more packages. Caller sends rtp as usual.

    – second ReInvite by UAS (0.69s after first ReInivte)
    – doesn’t contain the c field
    – m= audio 0 RTP/AVP 8 -> audio is stopped (why!?)
    – 488 (Not acceptable here) by UAC (log entry: Insufficient information
    in SDP (c=))
    – ACK by UAS

    about 2s later the same ReInivte w/o c-field can be seen – procedure as described for second ReInvite.

    – about 9s after the third ReInivte procedure, 3 option packages
    arrive and are acked (200) within 0.01s.

    – asterisk ends call by sending bye because of 31s missing rtp stream
    by UAS.

    **********************************************
    * *
    * BIG FAT QUESTION: *
    * Why does UAS stop the rtp stream? *
    * *
    **********************************************

  • Meanwhile I “solved” the problem by switching to pjsip. Using pjsip, the behavior of session-timers is default (