Hi,
I am running asterisk 1.8.5.0 and have compiled in the srtp module
All but Snom phones are working.
I have set the srtp tag on the snoms to 80 and RTP/SAVP to mandatory and they
worked for a few hours. This morning all snoms are reporting this when trying
to make a call (this is snom calling snom).
-snip--
== Using SIP RTP CoS mark 5
-- Executing [1@default-outbound08:1] Dial("SIP/10002-0012",
"SIP/1,30") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/1
-- SIP/1-0013 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [1@default-outbound08:2] VoiceMail("SIP/10002-0012",
"1,uj") in new stack
[Aug 3 11:58:29] WARNING[9543]: res_srtp.c:384 ast_srtp_unprotect: SRTP
unprotect: authentication failure
[Aug 3 11:58:29] WARNING[9543]: res_srtp.c:384 ast_srtp_unprotect: SRTP
unprotect: authentication failure
-- Playing 'vm-theperson.g729' (language 'en')
-- Playing 'digits/1.g729' (language 'en')
-- Playing 'digits/0.g729' (language 'en')
-- Playing 'digits/0.g729' (language 'en')
-- Playing 'digits/0.g729' (language 'en')
-- Playing 'digits/0.g729' (language 'en')
sage*CLI>
Disconnected from Asterisk server
[root@sage asterisk]#
---snip---
The interesting thing here is the call fails at this point and for some reason
the cli disconnects when the call fails.
Here is a call to a mobile which connects but the call dies in about 4 seconds
--snip
== Using SIP RTP CoS mark 5
-- Executing [0429835743@default-outbound08:1] Dial("SIP/10002-",
"SIP/private-sip/0429835743") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/private-sip/0429835743
-- SIP/private-sip-0001 is ringing
-- SIP/private-sip-0001 answered SIP/10002-
[Aug 3 12:06:05] WARNING[10146]: res_srtp.c:384 ast_srtp_unprotect: SRTP
unprotect: authentication failure
[Aug 3 12:06:05] WARNING[10146]: res_srtp.c:384 ast_srtp_unprotect: SRTP
unprotect: authentication failure
sage*CLI>
Disconnected from Asterisk server
--snip
I have done heaps of reading on SRTP unprotect error but cant really work it
out from that.
Q. should I try the patch mentioned below and forget about snoms doing 80 bit
incription or should I persevere with making this work?
thanks James
---snip---
Patch SRTP for 32bit
SRTP have a cryptographic hash to check the integrity of the encrypted packets.
It support two hash size:
● 32bit
● 80bit
In order to properly fine tune SRTP for mobile networks and to have
compatibility with PrivateGSM Enterprise we must use
SRTP with hash at 32bit (HMAC_SHA1_32).
Asterisk 1.8 by default does not announce in SDP both 32bit and 80bit, but only
the 80bit version even if both are supported.
This very small 1 line patch make Asterisk by default work with SRTP hash at
32bit .
Download the patch for HMAC_SHA1_32 RTP crypto offer
48. wget
http://sourceforge.net/projects/Asterisk-amr/files/1.8.0-rc2_crypto_offer.diff/download
Apply the patch
49. cd Asterisk-1.8.0/ && patch -p2 < ../1.8.0-rc2_crypto_offer.diff
Go to Asterisk-1.8.0/ folder50. cd ..
Recompile Asterisk ,
51. make ; make instal
snip--
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