Hi,<br><br>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: <br>
<br>[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<br>[Oct 2 10:59:48] NOTICE[24868]: sdp_crypto.c:242 sdp_crypto_process: SRTP crypto offer not acceptable<br>
[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<br>[Oct 2 10:59:48] NOTICE[24868]: sdp_crypto.c:242 sdp_crypto_process: SRTP crypto offer not acceptable<br>
[Oct 2 10:59:48] WARNING[24868]: chan_sip.c:7939 process_sdp: Can't provide secure audio requested in SDP offer<br><br>What means this?<br><br>By debugging sip messages: <br><br><--- SIP read from TLS:UA_IP_ADDRESS:60415 ---><br>
INVITE sips:202@AST_IP_ADDRESS;user=phone SIP/2.0<br>Via: SIP/2.0/TLS 192.168.105.199:5051;branch=z9hG4bKac781732149;alias<br>Max-Forwards: 70<br>From: "201" <sips:201@sdft;user=phone>;tag=1c781729204<br>To: <sips:202@AST_IP_ADDRESS;user=phone><br>
Call-ID: <a href="mailto:781728720312000192946@192.168.105.199">781728720312000192946@192.168.105.199</a><br>CSeq: 1 INVITE<br>Contact: <sips:201@192.168.105.199:5051;user=phone;transport=tls><br>Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat<br>
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE<br>User-Agent: Audiocodes-Sip-Gateway-Vox-Evo MP-118/v.5.60A.024.003<br>Content-Type: application/sdp<br>Content-Disposition: session<br>
Content-Length: 528<br><br>v=0<br>o=AudiocodesGW 781713142 781713021 IN IP4 192.168.105.199<br>s=Phone-Call<br>c=IN IP4 192.168.105.199<br>t=0 0<br>m=audio 6000 RTP/SAVP 0 8 18 4 96<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>
a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:4 G723/8000<br>a=fmtp:4 annexa=no<br>a=rtpmap:96 telephone-event/8000<br>a=fmtp:96 0-15<br>a=ptime:20<br>a=sendrecv<br>a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:EFG/GFJBnNMdfJ2/hBCyJmgdPS6MNkuOscQEJR3E|2^31<br>
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:IdPMZ5yfypyQPt2q0HPYnVojTSWj1el7cOB6LOEq|2^31<br><br><br><-------------><br>--- (14 headers 19 lines) ---<br>Using INVITE request as basis request - <a href="mailto:781728720312000192946@192.168.105.199">781728720312000192946@192.168.105.199</a><br>
Found peer '201' for '201' from UA_IP_ADDRESS:60415<br>sbc06*CLI><br><--- Reliably Transmitting (NAT) to UA_IP_ADDRESS:60415 ---><br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/TLS 192.168.105.199:5051;branch=z9hG4bKac781732149;alias;received=UA_IP_ADDRESS<br>
From: "201" <sips:201@sdft;user=phone>;tag=1c781729204<br>To: <sips:202@AST_IP_ADDRESS;user=phone>;tag=as1bf72d42<br>Call-ID: <a href="mailto:781728720312000192946@192.168.105.199">781728720312000192946@192.168.105.199</a><br>
CSeq: 1 INVITE<br>Server: Asterisk PBX SVN-group-srtp-r183146-/trunk<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces, timer<br>WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="526064ea"<br>
Content-Length: 0<br><br><br><------------><br>Scheduling destruction of SIP dialog '<a href="mailto:781728720312000192946@192.168.105.199">781728720312000192946@192.168.105.199</a>' in 32000 ms (Method: INVITE)<br>
sbc06*CLI><br><--- SIP read from TLS:UA_IP_ADDRESS:60415 ---><br>ACK sips:202@AST_IP_ADDRESS;user=phone SIP/2.0<br>Via: SIP/2.0/TLS 192.168.105.199:5051;branch=z9hG4bKac781732149;alias<br>Max-Forwards: 70<br>From: "201" <sips:201@sdft;user=phone>;tag=1c781729204<br>
To: <sips:202@AST_IP_ADDRESS;user=phone>;tag=as1bf72d42<br>Call-ID: <a href="mailto:781728720312000192946@192.168.105.199">781728720312000192946@192.168.105.199</a><br>CSeq: 1 ACK<br>Contact: <sips:201@192.168.105.199:5051;user=phone;transport=tls><br>
Supported: em,timer,replaces,path,early-session,resource-priority<br>Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE<br>User-Agent: Audiocodes-Sip-Gateway-Vox-Evo MP-118/v.5.60A.024.003<br>
Content-Length: 0<br><br><br><-------------><br>--- (12 headers 0 lines) ---<br>sbc06*CLI><br><--- SIP read from TLS:UA_IP_ADDRESS:60415 ---><br>INVITE sips:202@AST_IP_ADDRESS;user=phone SIP/2.0<br>Via: SIP/2.0/TLS 192.168.105.199:5051;branch=z9hG4bKac781931225;alias<br>
Max-Forwards: 70<br>From: "201" <sips:201@sdft;user=phone>;tag=1c781729204<br>To: <sips:202@AST_IP_ADDRESS;user=phone><br>Call-ID: <a href="mailto:781728720312000192946@192.168.105.199">781728720312000192946@192.168.105.199</a><br>
CSeq: 2 INVITE<br>Authorization: Digest username="201",realm="asterisk",nonce="526064ea",uri="sips:202@AST_IP_ADDRESS",algorithm=MD5,response="64f012c1334a4eb355f256c2569c61f6"<br>
Contact: <sips:201@192.168.105.199:5051;user=phone;transport=tls><br>Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat<br>Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE<br>
User-Agent: Audiocodes-Sip-Gateway-Vox-Evo MP-118/v.5.60A.024.003<br>Content-Type: application/sdp<br>Content-Disposition: session<br>Content-Length: 528<br><br>v=0<br>o=AudiocodesGW 781713142 781713021 IN IP4 192.168.105.199<br>
s=Phone-Call<br>c=IN IP4 192.168.105.199<br>t=0 0<br>m=audio 6000 RTP/SAVP 0 8 18 4 96<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:4 G723/8000<br>a=fmtp:4 annexa=no<br>
a=rtpmap:96 telephone-event/8000<br>a=fmtp:96 0-15<br>a=ptime:20<br>a=sendrecv<br>a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:EFG/GFJBnNMdfJ2/hBCyJmgdPS6MNkuOscQEJR3E|2^31<br>a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:IdPMZ5yfypyQPt2q0HPYnVojTSWj1el7cOB6LOEq|2^31<br>
<br><br><-------------><br>--- (15 headers 19 lines) ---<br>Sending to UA_IP_ADDRESS : 60415 (NAT)<br>Using INVITE request as basis request - <a href="mailto:781728720312000192946@192.168.105.199">781728720312000192946@192.168.105.199</a><br>
Found peer '201' for '201' from UA_IP_ADDRESS:60415<br>Found RTP audio format 0<br>Found RTP audio format 8<br>Found RTP audio format 18<br>Found RTP audio format 4<br>Found RTP audio format 96<br>Peer audio RTP is at port <a href="http://192.168.105.199:6000">192.168.105.199:6000</a><br>
Found audio description format PCMU for ID 0<br>Found audio description format PCMA for ID 8<br>Found audio description format G729 for ID 18<br>Found audio description format G723 for ID 4<br>Found audio description format telephone-event for ID 96<br>
[Oct 2 08:37:48] NOTICE[23034]: sdp_crypto.c:232 sdp_crypto_process: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80 inline:EFG/GFJBnNMdfJ2/hBCyJmgdPS6MNkuOscQEJR3E|2^31<br>[Oct 2 08:37:48] NOTICE[23034]: sdp_crypto.c:242 sdp_crypto_process: SRTP crypto offer not acceptable<br>
[Oct 2 08:37:48] NOTICE[23034]: sdp_crypto.c:232 sdp_crypto_process: Crypto life time unsupported: crypto:2 AES_CM_128_HMAC_SHA1_32 inline:IdPMZ5yfypyQPt2q0HPYnVojTSWj1el7cOB6LOEq|2^31<br>[Oct 2 08:37:48] NOTICE[23034]: sdp_crypto.c:242 sdp_crypto_process: SRTP crypto offer not acceptable<br>
[Oct 2 08:37:48] WARNING[23034]: chan_sip.c:7939 process_sdp: Can't provide secure audio requested in SDP offer<br>sbc06*CLI><br><--- Reliably Transmitting (NAT) to UA_IP_ADDRESS:60415 ---><br>SIP/2.0 488 Not acceptable here<br>
Via: SIP/2.0/TLS 192.168.105.199:5051;branch=z9hG4bKac781931225;alias;received=UA_IP_ADDRESS<br>From: "201" <sips:201@sdft;user=phone>;tag=1c781729204<br>To: <sips:202@AST_IP_ADDRESS;user=phone>;tag=as1bf72d42<br>
Call-ID: <a href="mailto:781728720312000192946@192.168.105.199">781728720312000192946@192.168.105.199</a><br>CSeq: 2 INVITE<br>Server: Asterisk PBX SVN-group-srtp-r183146-/trunk<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces, timer<br>Content-Length: 0<br><br><br><------------><br>Scheduling destruction of SIP dialog '<a href="mailto:781728720312000192946@192.168.105.199">781728720312000192946@192.168.105.199</a>' in 32000 ms (Method: INVITE)<br>
sbc06*CLI><br><--- SIP read from TLS:UA_IP_ADDRESS:60415 ---><br>ACK sips:202@AST_IP_ADDRESS;user=phone SIP/2.0<br>Via: SIP/2.0/TLS 192.168.105.199:5051;branch=z9hG4bKac781931225;alias<br>Max-Forwards: 70<br>From: "201" <sips:201@sdft;user=phone>;tag=1c781729204<br>
To: <sips:202@AST_IP_ADDRESS;user=phone>;tag=as1bf72d42<br>Call-ID: <a href="mailto:781728720312000192946@192.168.105.199">781728720312000192946@192.168.105.199</a><br>CSeq: 2 ACK<br>Contact: <sips:201@192.168.105.199:5051;user=phone;transport=tls><br>
Supported: em,timer,replaces,path,early-session,resource-priority<br>Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE<br>User-Agent: Audiocodes-Sip-Gateway-Vox-Evo MP-118/v.5.60A.024.003<br>
Content-Length: 0<br><br><br>Thanks in advance<br><br>Szasz Szabolcs<br>