search for: sdp_crypto

Hi, I have set up an asterisk with TLS and SRTP support. The SRTP is working with Phonerlite softphone. I have problem with the SRTP, when I make calls on Audiocodes gateway . I got the folloowing messages on asterisk: [Oct 2 10:59:48] NOTICE[24868]: sdp_crypto.c:232 sdp_crypto_process: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80 inline:SL+jOTOj8J1jTFgC+ETx5ORfFEWB5kxk5Ysr0XcI|2^31 [Oct 2 10:59:48] NOTICE[24868]: sdp_crypto.c:242 sdp_crypto_process: SRTP crypto offer not acceptable [Oct 2 10:59:48] NOTICE[24868]: sdp_crypto.c:232 sdp_...

Hello, I have added the following to the peer definition : ignorecryptolifetime=yes But still Asterisk tells me : [Oct 9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:244 sdp_crypto_process: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ikW6yFvdVkSaeTuVO1isTQkdaxOjgQjMEMSGUf+K|2^32 [Oct 9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:254 sdp_crypto_process: SRTP crypto offer not acceptable [Oct 9 14:02:34] WARNING[31980]: chan_sip.c:91...

...vf-v0 m=audio 50096 RTP/SAVP 0 18 120 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:120 telephone-event/8000 a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:zUVSWsFB/WjVtLxXojBT7zbNvuQ4BkOwcCkD/AjM|2^20 UNENCRYPTED_SRTCP And on CLI I see, DEBUG[1568][C-00000000] sip/sdp_crypto.c: local_key64 7vXot5kn/sl/GYv5ENN6yW0PZZapQ00c++biLgoX len 40 WARNING[1568][C-00000000] sip/sdp_crypto.c: Unsupported crypto parameters: UNENCRYPTED_SRTCP DEBUG[1568][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:zUVSWsFB/WjVtLxXojBT7zbNvuQ4Bk...

...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...

Hi all, i try to deactivate SRTP in asterisk 11. In sip.conf: tlsenable=no encryption=no transport=udp srtpcapable=no but when I try to make a call comes following message: [May 9 15:19:03] DEBUG[24745][C-00000086]: sip/sdp_crypto.c:285 sdp_crypto_process: Accepting crypto tag 1 [May 9 15:19:03] DEBUG[24745][C-00000086]: sip/sdp_crypto.c:310 sdp_crypto_offer: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:SYZjzhCe4mg0M18YvnkqtrH9lD3+/LQb3PuMoOI0 [May 9 15:19:03] WARNING[24745][C-00000086]: chan_sip.c:10526 process_...

...at shutdown (Reported by Corey Farrell) * ASTERISK-23814 - No call started after peer dialed (Reported by Igor Goncharovsky) * ASTERISK-23673 - Security: DOS by consuming the number of allowed HTTP connections. (Reported by Richard Mudgett) * ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite a DEBUG level of zero (Reported by Rusty Newton) * ASTERISK-23766 - [patch] Specify timeout for database write in SQLite (Reported by Igor Goncharovsky) * ASTERISK-23818 - PBX_Lua: after asterisk startup module is loaded, but dialplan not available (Reported by...

...at shutdown (Reported by Corey Farrell) * ASTERISK-23814 - No call started after peer dialed (Reported by Igor Goncharovsky) * ASTERISK-23673 - Security: DOS by consuming the number of allowed HTTP connections. (Reported by Richard Mudgett) * ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite a DEBUG level of zero (Reported by Rusty Newton) * ASTERISK-23766 - [patch] Specify timeout for database write in SQLite (Reported by Igor Goncharovsky) * ASTERISK-23818 - PBX_Lua: after asterisk startup module is loaded, but dialplan not available (Reported by...

...y Corey Farrell) * ASTERISK-23609 - Security: AMI action MixMonitor allows arbitrary programs to be run (Reported by Corey Farrell) * ASTERISK-23673 - Security: DOS by consuming the number of allowed HTTP connections. (Reported by Richard Mudgett) * ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite a DEBUG level of zero (Reported by Rusty Newton) * ASTERISK-23766 - [patch] Specify timeout for database write in SQLite (Reported by Igor Goncharovsky) * ASTERISK-23844 - Load of pbx_lua fails on sample extensions.lua with Lua 5.2 or greater due to addition of...

...y Corey Farrell) * ASTERISK-23609 - Security: AMI action MixMonitor allows arbitrary programs to be run (Reported by Corey Farrell) * ASTERISK-23673 - Security: DOS by consuming the number of allowed HTTP connections. (Reported by Richard Mudgett) * ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite a DEBUG level of zero (Reported by Rusty Newton) * ASTERISK-23766 - [patch] Specify timeout for database write in SQLite (Reported by Igor Goncharovsky) * ASTERISK-23844 - Load of pbx_lua fails on sample extensions.lua with Lua 5.2 or greater due to addition of...

...al' == Spawn extension (from-internal-xfer, 41720, 1) exited non-zero on 'DAHDI/1-1' -- Executing [41720 at from-internal-xfer:1] Set("DAHDI/1-1", "__RINGTIMER=20") in new stack And finally answered on 41720 [2013-07-03 13:43:16] DEBUG[29747][C-00004685]: sip/sdp_crypto.c:310 sdp_crypto_offer: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:5D038u88tI6PLyruDovyQIku9PH7exEAL3Qolc9m == Extension Changed 41720[ext-local] new state InUse for Notify User 41711 == Extension Changed 41720[ext-local] new state InUse for Notify User 41715 -- SIP/41720-000001...

I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regi?es: (11)4063-6100