[asterisk-bugs] [JIRA] Created: (ASTERISK-20233) SRTP not working with some devices (Eg Grandstream gxv3175) - Message "Can't provide secure audio requested in SDP offer"
tootai (JIRA)
noreply at issues.asterisk.org
Wed Aug 15 08:59:07 CDT 2012
SRTP not working with some devices (Eg Grandstream gxv3175) - Message "Can't provide secure audio requested in SDP offer"
-------------------------------------------------------------------------------------------------------------------------
Key: ASTERISK-20233
URL: https://issues.asterisk.org/jira/browse/ASTERISK-20233
Project: Asterisk
Issue Type: Bug
Security Level: None
Components: Channels/chan_sip/SRTP
Affects Versions: 10.7.0
Environment: RHEL5 Linux 2.6.18-308.11.1.el5
Reporter: tootai
Here is output
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 at 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
[...]
Call is ended with 488 error.
Same setup with blink softphone is OK. Difference is
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:j8ZJtC63YQGWyCMspHXEL6ca9VsuPcc2OBJk+Qav
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:yxZP4ZTdJa8vcEV4FVQrkk/s7LLkjXlHeNzkCWWv
So could the "|2^31" at the end of the crypto line be the cause ...
--
Daniel
--
This message is automatically generated by JIRA.
For more information on JIRA, see: http://www.atlassian.com/software/jira
More information about the asterisk-bugs
mailing list