Hello!
I'm facing the problem, that *sometimes* the SDP session ID isn't incremented
in the 200 OK, which asterisk sends as answer to a ReInvite it got from the
peer (use case: session
timer handling). This leads to broken calls, because the SDP session ID must be
incremented if the session description has changed (the session description has
changed).
Modifying the SDP session ID is possible in
res/res_pjsip/pjsip_message_filter.c / filter_on_tx_message() for SDPs
contained in an Invite, which is created and sent by asterisk.
At the moment, I'm already modifying the SDP session ID at this place, because
of another problem:
--- asterisk-16.5.0/res/res_pjsip/pjsip_message_filter.c 2019-04-04
16:49:57.000000000 +0200
+++ asterisk-16.5.0/res/res_pjsip/pjsip_message_filter.c 2019-06-18
14:46:57.784000000 +0200
@@ -340,6 +340,23 @@
}
}
+ /* set new sdp version */
+ ast_debug(5, "increment SDP version, starting from
%x\n",sdp->origin.version);
+ time_t sec;
+ time_t cseqt=1;
+ sec = time(NULL);
+ cseq = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CSEQ, NULL);
+ if (cseq) {
+ cseqt=cseq->cseq;
+ }
+ if (sec + cseqt <= sdp->origin.version) {
+ /* actual time can be same if sdp is generated more often
than once a second */
+ sdp->origin.version = sec + cseqt + (sdp->origin.version -
(sec + cseqt)) + 1;
+ }
+ else {
+ sdp->origin.version=sec + cseqt;
+ }
+
pjsip_tx_data_invalidate_msg(tdata);
}
Maybe it's too late and the processing of the 200 OK doesn't hit this part at
all?
Or is there any other possibility to modify the SDP session ID contained in the
200 OK, that is sent by asterisk as an answer to a ReInvite?
Thanks
Michael
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
