Hello, Sorry for the confusion. The point is the following: 1) call is setup correctly 2) just after one side send a RE-INVITE with a new port and new codec. 3) The other leg reject the RE-INVITE with a 415
During the RE-INVITE, the first RTP relay on RTPengine is removed (with a delay of 30 sec). As the RE-INVITE with the new port is rejected the current RTP relay is still used, so we must not delete them. BR Laurent -----Original Message----- From: sr-users <sr-users-boun...@lists.kamailio.org> On Behalf Of Daniel-Constantin Mierla Sent: jeudi, 5 décembre 2019 11:46 To: sr-users@lists.kamailio.org; Daniel Tryba <d.tr...@pocos.nl> Subject: Re: [SR-Users] RTPengine with rejected RE-INVITE when port is changing On 05.12.19 11:33, Daniel Tryba wrote: > On Thu, Dec 05, 2019 at 09:37:51AM +0000, Laurent Schweizer wrote: >> Hello, >> >> I already see old post about this : >> https://opensips.org/pipermail/users/2014-November/030451.html >> >> but I???m interested to know if now they is a solution ???? >> >> so the issue is a RE-INVITE rejected (415 Unsupported Media or 488 ) and the >> RTP port is changing. >> Is that case the old RTP relay (I???m using RTPengine) is destroyed after >> 30sec. >> Any way to restore them ? > I'm confused: > - opensips != kamailio > - rtpproxy != rtpengine > > But RFC 3261 states (page 16): > > During the session, either Alice or Bob may decide to change the > characteristics of the media session. This is accomplished by > sending a re-INVITE containing a new media description. This re- > INVITE references the existing dialog so that the other party knows > that it is to modify an existing session instead of establishing a > new session. The other party sends a 200 (OK) to accept the change. > The requestor responds to the 200 (OK) with an ACK. If the other > party does not accept the change, he sends an error response such as > 488 (Not Acceptable Here), which also receives an ACK. However, the > failure of the re-INVITE does not cause the existing call to fail - > the session continues using the previously negotiated > characteristics. Full details on session modification are in Section > 14. > > So if this happens with kamailio/rtpengine that is a bug in (I presume) > rtpengine. RTPEngine itself doesn't send SIP replies. Typically the 415 or 488 are sent by the endpoints. Kamailio doesn't do it by default, but of course can do it if explicitely wanted in the kamailio .cfg routing actions. My guess is that the endpoint sends the 415/488, so all this doesn't have to do anything with rtpengine/kamailio -- that can be checked by looking at the sip network traffic on kamailio server to see where 415/488 is coming from. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
