search for: aes_cm_128_hmac_sha1_32

...: 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:TyBSx7QAdczhqkuh+/eK2dWEH3c9sq7qa8r9FycS|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] WARNING[24868]: chan_sip.c:7939 process_sdp: Can't provide secure audio requested in SDP offer What means this? By debu...

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

...y 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: *** Our native formats are 0x100 (g729) [2010-12-23 11:06:22] DEBUG[5941] chan_sip.c: *** Joint capabilities are 0x100 (g729) [2010-12-23 11:06:22] DEBUG[5941] chan_sip.c: *** Our capabilities a...

...AYGl a=ice-pwd:3tLKvT97tf0GQr+e8v8bKncd a=ice-options:google-ice a=fingerprint:sha-256 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=rtp...

...talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 473 v=0 o=root 1775117380 1775117380 IN IP4 172.28.20.4 s=call c=IN IP4 172.28.20.4 t=0 0 m=audio 59502 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:8TM5Qape/dsy05garT9U1EHtmdo6iXy52b2Dc/Kc a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv --- (1...

...=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;alias;received=my.pub.lic.ip;rport=53416 From: <sip:testacc77005 at ast.ser.ver.ip:5061&...

...ransport=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_sdp: We are requesting SRTP for audio, but they responded without it! (call from a snom) I understand this message so that asterisk continue attempts to use SRTP. I could disable...

...pwd:kQ91vFMHVr2lOkZfjGDLSfO+ ????a=ice-options:google-ice ????a=fingerprint:sha-256 81: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 ?...

...cation/sdp > Content-Length: 522 > P-Asserted-Identity: <sip:<CALLING_PARTY_PHONE_NUMBER>@sipgate.de> > > v=0 > o=root 269390684 269390684 IN IP4 192.168.0.8 > s=call > c=IN IP4 217.10.77.20 > t=0 0 > m=audio 62652 RTP/AVP 9 0 8 3 99 108 18 101 > a=crypto:1 AES_CM_128_HMAC_SHA1_32 > inline:Ed8iHaP3BXNVeXHj98PRa6sJyImNer3ImjUvDZps > a=rtpmap:9 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:99 G726-32/8000 > a=rtpmap:108 AAL2-G726-32/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telepho...

I'm in the process of testing a TLS/SRTP install. My experience is improving with each new challenge, but this one is a great test of my 2 month experience with Asterisk. When I dial 6003 from 6001, it takes 35 seconds until I get the error message that 6003 is circuit-busy. Any help would greatly be appreciated. Below is the error message and the extensions and sip.conf files. *CLI>