I am not sure if this is the right forum where i should post this bug, But anyways, i have Google Chrome stable version 20.0.1132.57 and when i try to send call through webrtc to sip gateway to an Asterisk box, it complaints unsupported crypto parameters, e.g.
[Jul 21 13:16:54] WARNING[1877]: sip/sdp_crypto.c:220 sdp_crypto_process: Unsupported crypto parameters:
Here are the lines from SDP Asterisk receives,
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:3rV4ghgPu870biKKPnc3lHK7ybhAEGDxnwKQu1vD
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:veqvQQbIBDpoK2peHhBIxsNkeeBjMYQ/aDNPXHQZ
It took me a while to figure out the problem, that there is an extra space before the end of each line, which makes asterisk think that there are more crypto parameters to parse, but it finds none, thus it complains for unsupported crypto parameter and rejects the call with 488 Not Acceptable Here.
If in my WebRTC to SIP gateway, i forcefully check and remove this extra space in a=crypto line then SRTP is successfully negotiated and cal establishes. Though there is no audio on either end, but that's a different problem i have yet to figure out.
Thank you.
--
Muhammad Shahzad
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CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN:
shari...@hotmail.comEmail:
shahe...@googlemail.com