[asterisk-dev] Asterisk 11.3.0-rc1 and srtp - white noise only

Martin Koenig koenig at starface.de
Wed Mar 20 10:26:12 CDT 2013


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:
 
Is this something that is known already, or has happened to someone else? If not, I´d move forward with a bug report.
 
Many thanks and best regards,
Martin
-- 
Martin König
Manager Presales & Quality Assurance
 
STARFACE GmbH
Stephanienstr. 102
76133 Karlsruhe
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20130320/a3471cc9/attachment-0001.htm>


More information about the asterisk-dev mailing list