[asterisk-bugs] [JIRA] (ASTERISK-21323) Asterisk 11 svn branch and srtp - white noise only

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21323?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=204611#comment-204611 ] 

Matt Jordan commented on ASTERISK-21323:
----------------------------------------

We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Please ensure that you have 'sip set debug on' enabled. Please also include your sip.conf file.
                
>  Asterisk 11 svn branch and srtp - white noise only
> ---------------------------------------------------
>
>                 Key: ASTERISK-21323
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21323
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/SRTP
>    Affects Versions: SVN, 11.3.0
>         Environment: ubuntu 12.10 / 12.04 64bit
>            Reporter: andrea
>            Severity: Critical
>
> Hi all,
>  
> we have just updated our Asterisk 11 testbed from 11.2.1 to 11.3.0-rc1. Now we notice that all sRTP calls fail with “white noise” in the media channel phone > asterisk.
>  
> Example 1:
>  
> Snom w/ srtp > asterisk > Yealink w/ srtp
> Both ends hear “white noise”
>  
> Snom w/ srtp > asterisk > Gigaset w/o srtp
> Snom hears Gigaset, Gigaset hears white noise.
>  
> There have been no other changes to the setup, SIP.conf  specifies
>  
> transport=tls
> encryption=yes
>  
> for the sRTP phones.
>  
> Asterisk is linked to libsrtp 1.4.2.
>  
> Here is the log (that imho looks good):
>  
> [Mar 20 16:24:26] VERBOSE[13676][C-00000003] netsock2.c:   == Using SIP RTP CoS mark 5
> [Mar 20 16:24:26] DEBUG[13676][C-00000003] sip/sdp_crypto.c: Accepting crypto tag 1
> [Mar 20 16:24:26] DEBUG[13676][C-00000003] sip/sdp_crypto.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:N9srofPYzJbWC6bj7zQojDIl8abAlO34r5biUS38
> [Mar 20 16:24:26] VERBOSE[13700][C-00000003] pbx.c:     -- Executing [5 at local:1] Dial("SIP/snom360.2-00000006", "sip/1941.ylnkt32") in new stack
> [Mar 20 16:24:26] VERBOSE[13700][C-00000003] netsock2.c:   == Using SIP RTP CoS mark 5
> [Mar 20 16:24:26] DEBUG[13700][C-00000003] sip/sdp_crypto.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XiQI1ThhND6k+DNhlIeEXGzK2LY5/NF1y0K+YDfL
> [Mar 20 16:24:26] VERBOSE[13700][C-00000003] app_dial.c:     -- Called sip/1941.ylnkt32
> [Mar 20 16:24:27] VERBOSE[13700][C-00000003] app_dial.c:     -- SIP/1941.ylnkt32-00000007 is ringing
> [Mar 20 16:24:35] DEBUG[13681][C-00000003] sip/sdp_crypto.c: Accepting crypto tag 1
> [Mar 20 16:24:35] DEBUG[13681][C-00000003] sip/sdp_crypto.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XiQI1ThhND6k+DNhlIeEXGzK2LY5/NF1y0K+YDfL
> [Mar 20 16:24:35] VERBOSE[13700][C-00000003] app_dial.c:     -- SIP/1941.ylnkt32-00000007 answered SIP/snom360.2-00000006
> [Mar 20 16:24:35] DEBUG[13700][C-00000003] sip/sdp_crypto.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:N9srofPYzJbWC6bj7zQojDIl8abAlO34r5biUS38
> [Mar 20 16:24:35] VERBOSE[13700][C-00000003] res_rtp_asterisk.c:

--
This message is automatically generated by JIRA.
If you think it was sent incorrectly, please contact your JIRA administrators: https://issues.asterisk.org/jira/secure/ContactAdministrators!default.jspa
For more information on JIRA, see: http://www.atlassian.com/software/jira