Could not set SRTP policies

Asterisk Asterisk Support
1

I try to make the communication between web based SIP client and SIP softphone in asterisk 11.it is get connected but when i make call from Web based client it shows the error as below[Mar 5 12:12:06] WARNING[24847][C-00000000]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP policies [Mar 5 12:12:06] WARNING[24847][C-00000000]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP policies [Mar 5 12:12:06] WARNING[24847][C-00000000]: chan_sip.c:10487 process_sdp: Rejecting secure audio stream without encryption details: audio 53745 RTP/SAVPF 111 103 104 0 8 106 105 13 126
please help me to troubleshoot this error…thanx…plz reply.

2

Trace of SDP exchange, please.

3

thanx for the reply…basically i am new to asterisk… i dont know what is SDP Exchange…can you explore more about it?..so i provide it for debugging.

4

en.wikipedia.org/wiki/Session_De … n_Protocol

wiki.asterisk.org/wiki/display/ … nformation

5

hi David…thanx for the reply…the following is debug information shown by asterisk when i called from web based SIP client to another SIP softphone[Mar 6 14:09:07] VERBOSE[20490] asterisk.c: Asterisk Ready. [Mar 6 14:09:07] VERBOSE[20490] config.c: == Parsing '/etc/asterisk/cli.conf': Found [Mar 6 14:09:15] VERBOSE[20492] asterisk.c: -- Remote UNIX connection [Mar 6 14:09:27] VERBOSE[20603] res_http_websocket.c: == WebSocket connection from '202.141.151.100:58858' for protocol 'sip' accepted using version '13' [Mar 6 14:09:27] VERBOSE[20603] chan_sip.c: -- Registered SIP '1060' at 202.141.XX.XX:58858 [Mar 6 14:09:41] VERBOSE[20603][C-00000000] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 6 14:09:41] WARNING[20603][C-00000000] sip/sdp_crypto.c: Could not set SRTP policies [Mar 6 14:09:41] WARNING[20603][C-00000000] sip/sdp_crypto.c: Could not set SRTP policies [Mar 6 14:09:41] WARNING[20603][C-00000000] chan_sip.c: Rejecting secure audio stream without encryption details: audio 53760 RTP/SAVPF 111 103 104 0 8 106 105 13 126

6

yes…i solved the error after reinstalling the libsrtp and reconfigure the asterisk…