search for: aes_cm_128_hmac_sha1_80

...sdp? FYI SDP looks like this. v=0 o=- 1429194215 1 IN IP4 XX.XX.XX.XX s=- c=IN IP4 XX.XX.XX.XX b=TIAS:64000 t=0 0 a=avf:avc=n prio=n a=csup:avf-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...

...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_crypto_process: Crypto life time unsupported: crypto:2 AES_CM_128_HMAC_SHA1_32 inline:TyB...

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:9129 process_sdp: Can't provide secure audio requested in SDP offer What else do I nee...

Hi folks, At least several endpoints (soft phone and desk phones) are supporting various 256 bit ciphers for SRTP these days. I *believe* libsrtp has been updated to allow this, and that only the code in Asterisk has not been been updated to allow these stronger ciphers. Would anyone with the know-how be willing/able to submit a patch ? Thank you, Kevin Long

....168.1.104 s=SIP Call c=IN IP4 192.168.1.104 t=0 0 m=audio 5020 RTP/SAVP 0 8 18 9 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:8m7ZfG+0t3KBFGK40IfDO11SZ6D54glKKIwdgo00|2^32 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:nn+id/sSK7OErMfnZZduKNPLejpscxx1vUQB2seO|2^32 <--- Reliably Transmitting (NAT) to my.pub.lic.ip:53416 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.1.104:5068;branch=z9hG4bK60724585;al...

...o 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' ... [2010-12-23 11:06:22] DEBUG[5941] chan_sip.c:...

...56 89:D4:EB:6E:9C:41:36:03:A1:44:CD:A2:08:78:CD:86:FE:EC:30:09:53:0F:77:CE:BA:8E:DE:8C:1B:A1:41:10 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:dYMEPP1zoNS/W70Ln6cnBCtHXDCq6ciLZmHDHdFj a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Gr23SpFGDiukOKyrrfAauWssQ+3pYjD0jwkK9hOo a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60...

...Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: PBXe 1.4.0 Content-Type: application/sdp Content-Length: 342 v=0 o=- 772596305 772596305 IN IP4 192.168.1.4 s=Asterisk c=IN IP4 192.168.1.4 t=0 0 m=audio 14476 RTP/SAVP 0 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Ojz7o69EOsPsdsRTgNO/wtRWPsrWc2NSnOidNcqh a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv both phones SPA502, force_rport disabled for tls phone, here is my transports: [tls] type=transport ca_list_file=/pbx/k...

...DE:7E:6F:2B:88:8F:F3:30:82:92:DF:CB:FC:4B:63:BB:2E:BA:85:48:2B:B5:A6:C3:50:A1:42:E4:69:0E:91 ????a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level ????a=sendrecv ????a=mid:audio ????a=rtcp-mux ????a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:haR/UikskQr/AIrry5udqINI1hYfc5zY2I6jrkKT ????a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:waQfKIHI9UyjPVI0vrcUREDbSVZdtfCtRQK71/Ks ????a=rtpmap:111 opus/48000/2 ????a=fmtp:111 minptime=10 ????a=rtpmap:103 ISAC/16000 ????a=rtpmap:104 ISAC/32000 ????a=rtpmap:0 PCMU/8000 ????a=rtpmap:8 PCMA/8000 ????a=rtpmap:107 CN/48000 ????a=rtpmap:106 CN/32000 ????a=rtpmap:105 CN/16000 ????a=rtpm...

...179369454 cname:SvzCJjIAujxHGm9P a=ssrc:2179369454 msid:Kqg5QpXyqNeviT8qxUIRi8QNUaV7mUnFiDIF add6e533-c83d-42f2-b487-fcac8646ad32 a=ssrc:2179369454 mslabel:Kqg5QpXyqNeviT8qxUIRi8QNUaV7mUnFiDIF a=ssrc:2179369454 label:add6e533-c83d-42f2-b487-fcac8646ad32 a=sendrecv a=rtcp:11081 a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:RV9RgRP59zI6AoZKhGT4iq0Fj6A5tVbLy+zzj9JB a=setup:actpass a=fingerprint:sha-1 C2:D0:75:69:46:19:83:17:22:08:D4:8F:46:39:C7:AD:06:6A:CD:CC cheers, Olli -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachm...

...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/41720-0000016a answered DAHDI/1-1 It is evident from the trace that the context [se...

...; Server: Aastra 53i/2.1.0.2145 Supported: timer, replaces Content-Type: application/sdp Content-Length: 313 v=0 o=MxSIP 0 0 IN IP4 10.7.10.51 s=SIP Call c=IN IP4 10.7.10.51 t=0 0 m=audio 3000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ZXJ1UmhNLDFmQGNHYGAnRlpKbjEudk9Gfjh8blo/ a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (14 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.7.10.51:3000 Found audio description format PCMU for ID 0 Found audio descripti...

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