Summary: | ASTERISK-20233: SRTP not working with some devices (Eg Grandstream gxv3175) - Message "Can't provide secure audio requested in SDP offer" | ||||||||
Reporter: | tootai (tootai) | Labels: | |||||||
Date Opened: | 2012-08-15 08:57:23 | Date Closed: | 2012-08-15 17:22:59 | ||||||
Priority: | Major | Regression? | |||||||
Status: | Closed/Complete | Components: | Channels/chan_sip/SRTP | ||||||
Versions: | 10.7.0 | Frequency of Occurrence | Constant | ||||||
Related Issues: |
| ||||||||
Environment: | RHEL5 Linux 2.6.18-308.11.1.el5 | Attachments: | |||||||
Description: | Here is output
{noformat} v=0 o=<Private> 8001 8000 IN IP4 192.168.10.104 s=SIP Call c=IN IP4 192.168.10.104 t=0 0 m=audio 56008 RTP/SAVP 9 0 8 101 a=sendrecv a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:txJVSUpEW30QbN7XrYuyOgNHVOHBX4dshzqYUwzg|2^31 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:jcPz4ww9J2e7ONOZD4AkwronCQ8Jym6QNvXKz0jW|2^31 <-------------> [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: --- (17 headers 15 lines) --- [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Sending to 109.237.252.179:51974 (NAT) [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Using INVITE request as basis request - 633650172-36867-4@BJC.BGI.BA.BAE [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found peer '<private>' for '<private>' from xxx.xxx.xxx.xxx:51974 [2012-08-15 17:33:27] VERBOSE[9911] netsock2.c: == Using SIP RTP CoS mark 5 [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found RTP audio format 9 [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found RTP audio format 0 [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found RTP audio format 8 [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found RTP audio format 101 [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found audio description format G722 for ID 9 [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found audio description format PCMU for ID 0 [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found audio description format PCMA for ID 8 [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found audio description format telephone-event for ID 101 [2012-08-15 17:33:27] NOTICE[9911] sip/sdp_crypto.c: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80 inline:txJVSUpEW30QbN7XrYuyOgNHVOHBX4dshzqYUwzg|2^31 [2012-08-15 17:33:27] NOTICE[9911] sip/sdp_crypto.c: SRTP crypto offer not acceptable [2012-08-15 17:33:27] NOTICE[9911] sip/sdp_crypto.c: Crypto life time unsupported: crypto:2 AES_CM_128_HMAC_SHA1_32 inline:jcPz4ww9J2e7ONOZD4AkwronCQ8Jym6QNvXKz0jW|2^31 [2012-08-15 17:33:27] NOTICE[9911] sip/sdp_crypto.c: SRTP crypto offer not acceptable [2012-08-15 17:33:27] WARNING[9911] chan_sip.c: Can't provide secure audio requested in SDP offer [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: <--- Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:51974 ---> SIP/2.0 488 Not acceptable here [...] {noformat} Call is ended with 488 error. Same setup with blink softphone is OK. Difference is {noformat} a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:j8ZJtC63YQGWyCMspHXEL6ca9VsuPcc2OBJk+Qav a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:yxZP4ZTdJa8vcEV4FVQrkk/s7LLkjXlHeNzkCWWv {noformat} So could the {{|2^31}} at the end of the crypto line be the cause ... -- Daniel | ||||||||
Comments: | By: David Brillert (aragon) 2012-08-15 10:21:39.108-0500 I have the same problem with Audiocodes phones. Did some digging and found there was a patch for crypto key lifetime issues that you seem to be having and I was having as well. There is also an FAQ on the Grandstream website. http://www.grandstream.com/support/faq/gxp-enterprise-phone-series#faqGXP25 I tried the unsupported patch available at ASTERISK-17899 SRTP negotiation then works but I don't hear any audio (extension>server<extension) I can't get this SRTP stuff working... By: Matt Jordan (mjordan) 2012-08-15 17:22:41.647-0500 This is not a bug but a feature request. Asterisk at this time does not support lifetime for cryptographic keys (which is the part after the | in the key). As such, this issue will be closed out unless a patch is provided which actually adds support in Asterisk for lifetime. By: Matt Jordan (mjordan) 2012-08-15 17:22:53.876-0500 Features requests are no longer submitted to or accepted through the issue tracker. Features requests are openly discussed on the mailing lists [1] and Asterisk IRC channels and made note of by Bug Marshals. [1] http://www.asterisk.org/support/mailing-lists |