Hello! Ater several successful SRTP-enabled calls with SRTP set to Mandatory, asterisk starts to give the following warnings in Log:
WARNING[13714] res_srtp.c: SRTP unprotect: authentication failure (continiously) and client hears no sound. After i restart the client program it works fine again for a while. Then the same warning again. Asterisk 1.8.1.1, RealTime engine, sip peer has encrytion->yes The client program is CSipSimple on Android Here are some log file traces: Peer 0010101 is calling some number that is routed to context a2billing ... [2010-12-23 11:06:22] DEBUG[5941] sip/sdp_crypto.c: local_key64 3gWGFJAffj4Pn393BUPwe3/wOMx5/ndZyPtfno7L len 40 [2010-12-23 11:06:22] DEBUG[5941] sip/sdp_crypto.c: SRTP policy activated [2010-12-23 11:06:22] DEBUG[5941] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:0VyG/fnup0U9qDoTGlWvVuE5yAef5MfYU6F67oI+... OK. [2010-12-23 11:06:22] DEBUG[5941] chan_sip.c: We've already processed a crypto attribute, skipping 'crypto:2 AES_CM_128_HMAC_SHA1_32 inline:5X/Zqep5tNdDGFhOY1//VFQ7diCCH1Y1FUKgYXLp' ... 010-12-23 11:06:22] DEBUG[5941] chan_sip.c: build_route: Contact hop: <sip:0010...@78.84.207.114:5060;transport=UDP;ob> [2010-12-23 11:06:22] DEBUG[5941] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header. [2010-12-23 11:06:22] DEBUG[5941] chan_sip.c: Session-Expires: 1800 [2010-12-23 11:06:22] DEBUG[5941] chan_sip.c: Received Min-SE: 90 ... [2010-12-23 11:06:22] DEBUG[5931] chan_sip.c: -REALTIME- peer built. Name: 0010101. Peer objects: 660 [2010-12-23 11:06:22] DEBUG[5931] netsock2.c: Splitting '78.84.207.114' gives... [2010-12-23 11:06:22] DEBUG[5931] netsock2.c: ...host '78.84.207.114' and port '(null)'. [2010-12-23 11:06:22] DEBUG[5931] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [2010-12-23 11:06:22] DEBUG[5931] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [2010-12-23 11:06:22] DEBUG[5931] chan_sip.c: -REALTIME- loading peer from database to memory. Name: 0010101. Peer objects: 660 [2010-12-23 11:06:22] DEBUG[5931] chan_sip.c: Destroying SIP peer 0010101 [2010-12-23 11:06:22] DEBUG[5931] chan_sip.c: -REALTIME- peer Destroyed. Name: 0010101. Realtime Peer objects: 659 [2010-12-23 11:06:22] DEBUG[5931] devicestate.c: Changing state for SIP/0010101 - state 1 (Not in use) is this normal here? peer destroyed? [2010-12-23 11:06:22] DEBUG[5931] devicestate.c: device 'SIP/0010101' state '1' [2010-12-23 11:06:22] WARNING[8264] res_srtp.c: SRTP unprotect: authentication failure [2010-12-23 11:06:22] DEBUG[5941] chan_sip.c: = Looking for Call ID: 2WZXYS-qTPPfXylUor4tckg25TetmIVP (Checking From) --From tag 50FYKcXAUIrUwsIpR5xm9pjrSrMaDglb --To-tag as46be1cdb [2010-12-23 11:06:22] DEBUG[5941] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2010-12-23 11:06:22] DEBUG[5941] chan_sip.c: Stopping retransmission on '2WZXYS-qTPPfXylUor4tckg25TetmIVP' of Response 20465: Match Found [2010-12-23 11:06:22] WARNING[8264] res_srtp.c: SRTP unprotect: authentication failure [2010-12-23 11:06:22] WARNING[8264] res_srtp.c: SRTP unprotect: authentication failure Thanks in advance, Nick -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
