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<DIV><FONT size=2 face=Arial>Hi,</FONT></DIV>
<DIV><FONT size=2 face=Arial>I am running asterisk 1.8.5.0 and have compiled in
the srtp module</FONT></DIV>
<DIV><FONT size=2 face=Arial>All but Snom phones are working.</FONT></DIV>
<DIV><FONT size=2 face=Arial>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).</FONT></DIV>
<DIV><FONT size=2 face=Arial>---------snip------------------</FONT></DIV>
<DIV><FONT size=2 face=Arial> == Using SIP RTP CoS mark
5<BR> -- Executing [10000@default-outbound08:1]
Dial("SIP/10002-00000012", "SIP/10000,30") in new stack<BR> == Using SIP
RTP CoS mark 5<BR> -- Called SIP/10000<BR>
-- SIP/10000-00000013 is circuit-busy<BR> == Everyone is busy/congested at
this time (1:0/1/0)<BR> -- Executing
[10000@default-outbound08:2] VoiceMail("SIP/10002-00000012", "10000,uj") in new
stack<BR>[Aug 3 11:58:29] WARNING[9543]: res_srtp.c:384
ast_srtp_unprotect: SRTP unprotect: authentication failure<BR>[Aug 3
11:58:29] WARNING[9543]: res_srtp.c:384 ast_srtp_unprotect: SRTP unprotect:
authentication failure<BR> -- <SIP/10002-00000012>
Playing 'vm-theperson.g729' (language 'en')<BR> --
<SIP/10002-00000012> Playing 'digits/1.g729' (language
'en')<BR> -- <SIP/10002-00000012> Playing
'digits/0.g729' (language 'en')<BR> --
<SIP/10002-00000012> Playing 'digits/0.g729' (language
'en')<BR> -- <SIP/10002-00000012> Playing
'digits/0.g729' (language 'en')<BR> --
<SIP/10002-00000012> Playing 'digits/0.g729' (language
'en')<BR>sage*CLI><BR>Disconnected from Asterisk server<BR>[root@sage
asterisk]#</FONT></DIV>
<DIV><FONT size=2 face=Arial>-------snip-------<BR></DIV></FONT>
<DIV><FONT size=2 face=Arial>The interesting thing here is the call fails at
this point and for some reason the cli disconnects when the call
fails.</FONT></DIV>
<DIV><FONT size=2 face=Arial>Here is a call to a mobile which connects but the
call dies in about 4 seconds</FONT></DIV>
<DIV><FONT size=2 face=Arial>------snip--------</FONT></DIV>
<DIV><FONT size=2 face=Arial> == Using SIP RTP CoS mark
5<BR> -- Executing [0429835743@default-outbound08:1]
Dial("SIP/10002-00000000", "SIP/private-sip/0429835743") in new stack<BR>
== Using SIP RTP CoS mark 5<BR> -- Called
SIP/private-sip/0429835743<BR> -- SIP/private-sip-00000001 is
ringing<BR> -- SIP/private-sip-00000001 answered
SIP/10002-00000000<BR>[Aug 3 12:06:05] WARNING[10146]: res_srtp.c:384
ast_srtp_unprotect: SRTP unprotect: authentication failure<BR>[Aug 3
12:06:05] WARNING[10146]: res_srtp.c:384 ast_srtp_unprotect: SRTP unprotect:
authentication failure<BR>sage*CLI><BR>Disconnected from Asterisk
server<BR>------snip------------</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial>I have done heaps of reading on SRTP unprotect
error but cant really work it out from that.</FONT></DIV>
<DIV><FONT size=2 face=Arial>Q. should I try the patch mentioned below and
forget about snoms doing 80 bit incription or should I persevere with making
this work?</FONT></DIV>
<DIV><FONT size=2 face=Arial>thanks James</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial>---snip---</FONT></DIV>
<DIV><FONT size=2 face=Arial>Patch SRTP for 32bit<BR>SRTP have a cryptographic
hash to check the integrity of the encrypted packets.<BR>It support two hash
size:<BR>● 32bit<BR>● 80bit<BR>In order to properly fine tune SRTP for mobile
networks and to have compatibility with PrivateGSM Enterprise we must
use<BR>SRTP with hash at 32bit (HMAC_SHA1_32).<BR>Asterisk 1.8 by default does
not announce in SDP both 32bit and 80bit, but only the 80bit version even if
both are supported.<BR>This very small 1 line patch make Asterisk by default
work with SRTP hash at 32bit .<BR>Download the patch for HMAC_SHA1_32 RTP crypto
offer<BR>48. wget </FONT><A
href="http://sourceforge.net/projects/Asterisk-amr/files/1.8.0-rc2_crypto_offer.diff/download"><FONT
size=2
face=Arial>http://sourceforge.net/projects/Asterisk-amr/files/1.8.0-rc2_crypto_offer.diff/download</FONT></A><BR><FONT
size=2 face=Arial>Apply the patch<BR>49. cd Asterisk-1.8.0/ && patch -p2
< ../1.8.0-rc2_crypto_offer.diff<BR>Go to Asterisk-1.8.0/ folder50. cd
..<BR>Recompile Asterisk ,<BR>51. make ; make instal</FONT>
<DIV><FONT size=2
face=Arial>------------snip------------------</FONT></DIV></DIV>
<DIV><FONT size=2 face=Arial> </DIV></FONT></BODY></HTML>