Noisy sound static using tls betwean two phons yealink

hello friends, I am using tls and tcp in my configuration this is my
sip.conf

[general]
context=from-internal
srvlookup=yes
transport=tcp
tcpenable=yes
tcpbindaddr=0.0.0.0:5060
realm=192.168.1.123

transport=tls
tlsenable=yes
tlsbindaddr=0.0.0.0:5061
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlsprivatekey=/etc/asterisk/keys/asterisk.key
tlscafile=/etc/asterisk/keys/ca.crt
tlscapath=/etc/asterisk/keys
tlscipher=ALL
tlsclientmethod=tlsv1

[1001]
callerid=cliente1001<1001>
defaultuser=cliente1001
secret=123abc
host=dynamic
type=friend
context=from-internal
dtmfmode=rfc2833
transport=tls
disallow=all
allow=g726
allow=g722
allow=ulaw
allow=alaw
qualify=yes
directmedia=no
rtcp_mux=yes
encryption=yes

[1002]
callerid=cliente1002<1002>
defaultuser=cliente1002
secret=123abc
host=dynamic
type=friend
context=from-internal
dtmfmode=rfc2833
transport=tls
disallow=all
allow=g726
allow=g722
allow=ulaw
allow=alaw
qualify=yes
directmedia=no
rtcp_mux=yes
encryption=yes

extension.conf
[from-internal]
exten => _1XXX,1,Dial(SIP/${EXTEN})

The phones yealink are registered to the server with tls certificates and in the section advanced of yealink i selected RTP Encryption (SRTP) compulsory and in the sip.conf I add encryption=yes and when I make a call between the only two extents, I only listen to noise, this the same situation I configured in res_pjsip and I the call work very well there is no noise in the call.

This a image of a call

but if you pickup the phone you can hear only noise and I have tested with many codecs but the sound is only noise.

I hope somebody can help me.

If the noise is loud, the wrong encryption key is being used. That should be iimpossible within Asterisk.

thank you for your help, so you mean tlscipher=ALL? should I use another one?

I simply meant that the (session) encryption key used to encrypt was not the same as that used to decrypt. There should be no configuration option that could be used to cause this, and it is likely to be an implementation fault. As this is the first I’ve heard of it here, my presumption would be that the fault isn’t in Asterisk.

hello friend, but I did the same but in res_pjsip and it work without problem, I used the same process in both, in pjsip worked, in sip asterisk13 didnt work
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
with the asterisk13 I hear noisy and in asterisk16 using the same process I hear noisy.

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