This is SRTP library, http://srtp.sourceforge.net/srtp.html, version for SRTP - 1.4.4, adopts the AES encryption method, the RTP Packet is encrypted through RTSP Server to send out, after verification method is through the VLC Player, live555 RTSP Server is to use open source code,
Simple explain process, when there are wired to establish, in the live555 MultiFramedRTPSink this category for the following action:
(1) generate the key, use AES_BOX={xc3 xc1 0, 0 xee, 0, 0 x71, 0 x7d, 0 xa7, 0 x61, 0 x95, 0 XBB, 0 x87, 0 x85, 0 x78, 0 x79, 0 x0a, 0 xf7, 0 x1c, 0 x4e, 0 xe9, 0 xf8, 0 x59, 0 xe1, 0 x97, 0 xa4, 0 x14, 0 xa7, 0 x8d, 0 x5a, 0 XBC, 0 x74, 0 x51} (a total of 30 bytes), after a base64 encoding for
We7DcX2nYZW7h4V4eQr3HE7p + Fnhl6QUp41avHRR (a total of 40 bytes),
(2) the use of encryption, using crypto_policy_set_aes_cm_128_hmac_sha1_32 crypto_policy_set_rtcp_default () and (), at the same time with encryption and authentication,
(3) initializing the parameters of the need to use, malloc rtp_sender_ctx_t this structure, call srtp_create () hand in the key information,
(4) after completion of the above actions in the live555 sendPacketIfNecessary (), perform srtp_protect (), the RTP Header and RTP Packet to encrypt, finally provided by the RTSP Server send streaming to VLC Player,
The question now is, RTSP handshake messages go in front of (omitted)
V=0
O=1473678528576161 1 RTSP IN IP4 0.0.0.0
S=RTSP Server
T=0 0
A=range: NPT=0 -
A=control: *
M=0 video RTP/SAVP 99
A=crypto: 1 AES_CM_128_HMAC_SHA1_32 \ r \ ninline: we7DcX2nYZW7h4V4eQr3HE7p + Fnhl6QUp41avHRR \ r \ n
C=IN IP4 0.0.0.0
B=AS: 0
A=rtpmap: 99 H264/90000
A=FMTP: 99 packetization - mode=1; Profile - level - id=4 d002a; Sprop - parameter - sets=Z00AKpWoHgCJ + VA=, aO48gA==
A=control: track1
As mentioned in the RFC3711, m=video 0 RTP/SAVP 99, must use SAVP or SAVPF and want to join a=crypto... The inline:, but if you join in accordance with the above way, handshake process do the DESCRIBE command to end VLC no longer response, and to establish a connection,
Want to do,
(1) is the master key/AES_BOX content of salt use error?
(2) the Describe SDPLine write wrong?
(3) the VLC master key/salt to how to fill in? Multiplexer (tools -> preferences -> solution - RTP - SRTP key/salt)
(4) or other device can verify dial? (because of using Wireshark is unable to parse)
(5) or other places didn't notice?
Thanks for comment!
CodePudding user response:
You to ask, then a process can run? I was reading something SRTP,