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I have this chat system that's using SIP for voice transmission (no phones, only browser to browser in the same server). The sound goes just fine in both ends, but asterisk gives a warning twice (one for each client probably):

[Mar 11 09:01:27] WARNING[614][C-00000002]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10
[Mar 11 09:01:33] WARNING[614][C-00000002]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10

The keys where generated with:

./usr/src/asterisk-*/contrib/scripts/ast_tls_cert -C IpAddress -O "Company" -d /etc/asterisk/keys

sip.conf file:

[general]
udpbindaddr=0.0.0.0:5060
realm=IpAddress
transport=udp,ws,wss
videosupport=no
icesupport=yes
allowguest=no
allwaysreject=yes
rtptimeout=30

My peers are set in sip.conf as:

[000001]
host=dynamic
secret=000001
context=from-internal
type=friend
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw,alaw,gsm,all
dtlsenable=yes
dtlsverify=no
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass

And I'm using sipML5 for the client. Any insights to clear the warnings?

onebree
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Moisés
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  • I have the same issue. – Josh Jul 10 '15 at 16:16
  • Similar issue here. But for me, this warning message occured after the call was established (about 10+ seconds ago), and then MusicOnHold is played (only the callee can hear it), and then this warning message occured. When the call was established, the callee (a DAHDI channel) can hear the caller (WebRTC client, Firefox 40.0.4, jssip 0.7.4), but the caller can't hear the callee – LiuYan 刘研 Sep 10 '15 at 04:23
  • +1. Moreover I think this error causes ```[160280.900844] asterisk[11337]: segfault at 3c ip 00007f581e2f61b3 sp 00007f5805e89810 error 4 in libpjnath.so.2[7f581e2dc000+29000]``` segfault error. Asterisk starts to use 99% CPU and refuses to serve trunk connections. Have you guys solved this issue? – Alex Jul 11 '16 at 09:36

0 Answers0