On 12/27/2016 at 07:54 PM Michael Maier wrote: > 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.
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? * * * ********************************************** > 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 > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
