[asterisk-bugs] [Asterisk 0018674]: [patch] Unable to choose which SRTP suite to offer

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18674 
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Reported By:                bbeers
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18674
Category:                   Channels/chan_sip/SRTP
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                       SWP-3142 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 303637 
Request Review:              
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Date Submitted:             2011-01-25 09:56 CST
Last Modified:              2011-03-02 07:31 CST
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Summary:                    [patch] Unable to choose which SRTP suite to offer
Description: 
Setting encryption=yes in sip.conf will cause asterisk to
 generate a line in SIP INVITE SDP:

 a=crypto: AES_CM_128_HMAC_SHA1_80 ...

There is no way to specify that asterisk should offer
 AES_CM_128_HMAC_SHA1_32 instead of
 AES_CM_128_HMAC_SHA1_80.

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Relationships       ID      Summary
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has duplicate       0018893 Can't provide secure audio requested in...
related to          0018187 Indicate SRTP + Feature reqest
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 (0132515) keys (reporter) - 2011-03-02 07:31
 https://issues.asterisk.org/view.php?id=18674#c132515 
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I have been using 05.patch with encryption=aes_32. 
I am still getting these messages in log:


Outbound call from softphone TLS+SRTP (encryption=aes_32) to server A onto
server B using iax2:

chan_iax2.c:     -- Format for call is ulaw
app_dial.c:     -- IAX2/SSS-17853 is making progress passing it to
SIP/101-00000185
sip/sdp_crypto.c:   == Selecting 2 (AES_CM_128_HMAC_SHA1_32) for srtp
crypto offer.
res_srtp.c: SRTP unprotect: unsupported parameter
res_srtp.c: SRTP unprotect: authentication failure



Inbound call from server B to server A using iax2 and then onto softphone
using TLS+SRTP (encryption=aes_32):

netsock2.c:   == Using SIP RTP TOS bits 184
netsock2.c:   == Using SIP RTP CoS mark 5
chan_sip.c:   == Encrypted Media is required, offering suite 2
(AES_CM_128_HMAC_SHA1_32).
chan_sip.c:   == SRTP_CRYPTO_SUITE is set to 32
(AES_CM_128_HMAC_SHA1_32).
sip/sdp_crypto.c:   == Selecting 2 (AES_CM_128_HMAC_SHA1_32) for srtp
crypto offer.
app_dial.c:     -- Called 101
app_dial.c:     -- SIP/101-00000187 is ringing
res_rtp_asterisk.c: RTP Read too short
chan_sip.c:   == SRTP_CRYPTO_SUITE is set to 32
(AES_CM_128_HMAC_SHA1_32).
app_dial.c:     -- SIP/101-00000187 answered IAX2/701-11563
pbx.c:     -- Executing [s at macro-auto-blkvm:1] Set("SIP/101-00000187",
"__MACRO_RESULT=") in new stack
pbx.c:     -- Executing [s at macro-auto-blkvm:2] NoOp("SIP/101-00000187",
"Deleting: BLKVM/700/IAX2/701-11563 TRUE") in new stack
res_srtp.c: SRTP unprotect: authentication failure


Thanks for all your efforts bbeers. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-03-02 07:31 keys           Note Added: 0132515                          
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