[SR-Users] rtpengine UDP/TLS/RTP/SAVP <=> RTP/AVP re-INVITE

[Juha Heinanen]

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:jh at 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:jh at 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 at 192.98.102.30:5066;transport=tcp> from <sip:jh at 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:jh at 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?

-- juha