[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:all-tabpanel ]

Matt Jordan updated ASTERISK-21323:
-----------------------------------

    Description: 
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):

\[inline log removed\]


  was:
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:


    
>  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
>            Assignee: 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):
> \[inline log removed\]

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