On 11/17/2014 08:03 PM, Juha Heinanen wrote: > when i make call from UDP/TLS/RTP/SAVP baresip to RTP/AVP sems, > rtpengine gets called on initial invite/200 ok like this and audio works > fine: > > Nov 18 02:46:06 rautu /usr/bin/sip-proxy[926]: INFO: ===== > rtpengine_offer(ICE=force replace-session-connection replace-origin > via-branch=1 RTP/AVP trust-address) > > Nov 18 02:46:06 rautu /usr/bin/sip-proxy[878]: INFO: ===== > rtpengine_answer(ICE=force via-branch=2 trust-address) > > however, when baresip makes re-invite, rtpengine gets called using the > same offer/answer flags, but sems clears the call. on syslog, i see > this: > > Nov 18 02:46:59 rautu /usr/bin/sip-proxy[919]: INFO: Routing in-dialog INVITE > <sip:127.0.0.1:5090;transport=udp> from <sip:j...@test.tutpro.com> > Nov 18 02:46:59 rautu /usr/bin/sip-proxy[879]: INFO: ===== > rtpengine_answer(ICE=force via-branch=2 replace-session-connection > replace-origin) > Nov 18 02:46:59 rautu rtpengine[29718]: [abd2bcf75f71af57 port 50444] SRTP > output wanted, but no crypto suite was negotiated > Nov 18 02:46:59 rautu /usr/bin/sip-proxy[919]: INFO: Routing in-dialog ACK > <sip:127.0.0.1:5090;transport=udp> from <sip:j...@test.tutpro.com> > Nov 18 02:46:59 rautu rtpengine[29718]: [abd2bcf75f71af57 port 50462] > Discarded invalid SRTP packet: authentication failed > Nov 18 02:46:59 rautu sems[20439]: [#b7193b70] [receive, AmRtpAudio.cpp:212] > ERROR: decode() returned -4 > Nov 18 02:46:59 rautu /usr/bin/sip-proxy[878]: INFO: ===== rtpengine_delete() > Nov 18 02:46:59 rautu /usr/bin/sip-proxy[878]: INFO: Routing in-dialog BYE > <sip:jh-0x9a95b00@192.98.102.30:5066;transport=tcp> from > <sip:j...@as.test.tutpro.com> to <sip:192.98.102.30:46718;transport=TCP> > based on gruu > Nov 18 02:46:59 rautu rtpengine[29718]: [abd2bcf75f71af57 port 50463] Error > parsing RTCP header: invalid packet type > > and on baresip console this: > > dtls_srtp: ---> DTLS-SRTP complete (audio/RTCP) > Profile=AES_CM_128_HMAC_SHA1_80 > dtls_srtp: incoming DTLS connect from 192.98.102.30:50524 > dtls_srtp: verified sha-1 fingerprint OK > dtls_srtp: ---> DTLS-SRTP complete (audio/RTP) Profile=AES_CM_128_HMAC_SHA1_80 > srtp: recv: failed to decrypt RTCP-packet (Unknown error 217) > srtp: recv: failed to decrypt RTCP-packet (Unknown error 217) > srtp: recv: failed to decrypt RTP-packet (Unknown error 217) > srtp: recv: failed to decrypt RTP-packet (Unknown error 217) > srtp: recv: failed to decrypt RTP-packet (Unknown error 217) > srtp: recv: failed to decrypt RTP-packet (Unknown error 217) > sip:j...@as.test.tutpro.com: session closed: Connection reset by peer > > could it be that at some point during the re-invite, sems gets srtp audio > and therefore clears the call? if so, is it a bug in rtpengine?
Can you post the complete log from rtpengine for such a call and perhaps also make a pcap of the media packets? It's kinda hard to follow what exactly is going on with just what you posted. cheers _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
