Hi this is my first post , I’m new on this forum… I’m a newbie with asterisk and I have known asterisk 1 month ago …
I would configure asterisk with RSTP/TLS protocol , I started using asterisk tutorial SRTP (asterisk webdoc) and internet tutorials…but I received the same problem with all : when I answer to the call I can hear only noise, like an encryption call without decryption…
These are the tests that I have done with the same results :
loaded with VMWARE 7
CertOS 6.5 (32bit) -> asterisk sourcecode 1.8.5.0 + source code fix sdp_crypto.c + libsrtp installed
asterisk sourcecode 10.8.0 + libsrtp installed
Ubuntu 11.04 (32bit) -> asterisk bin version from repository 1.8.11.1digium
I have verified that res_srtp.o has been loaded (without errors) and I have started asterisk with
this command (asterisk -vvvvc).
I have tested these calls (100 -> 101 and 101 -> 100)
Normal Call UDP -> Call working well
TLS Call -> Call working well
SRPT(Mandatory) and TLS -> I can call, I can see TLS and SRTP call simbols but when I answer I can hear only noise
SRPT(Mandatory) without TLS -> I can call, I can see SRTP call simbol but when I answer I can hear only noise
SRPT(Optional) and TLS -> I couldn’t call and I receive from asterisk CLI WARNING[30156]: chan_sip.c:9457 process_sdp: We are requesting SRTP for audio, but they responded without it!
Software Used :
PC: Blink Phone Lite
IOS : Bria
Conf file changed :
sip.conf
[100]
type=peer
host=dynamic
secret=100
context=users
disallow=all
transport=udp, tls
port=5062
encryption=yes
allow=g722
[101]
type=peer
host=dynamic
secret=101
context=users
disallow=all
transport=udp, tls
port=5062
encryption=yes
allow=g722
[600]
type=peer
host=dynamic
secret=101
context=users
disallow=all
transport=udp,tls
port=5062
encryption=yes
allow=g722
extension.conf
[users]
; exten => 100,1,Set(_SIPSRTP_CRYPTO=enable)
exten => 100,1,Set(_SIP_SRTP_SDES=1)
exten => 100,2,Set(_SIPSRTP=enable)
exten => 100,3,Set(_SIPSRTP_CRYPTO=enable)
exten => 100,4,Dial(SIP/100)
;exten => 100,1,Set(CHANNEL(secure_bridge_signaling)=1)
; same => n,Set(CHANNEL(secure_bridge_media)=1)
; same => n,Dial(SIP/100)
;exten => 101,1,Set(CHANNEL(secure_bridge_signaling)=1)
; same => n,Set(CHANNEL(secure_bridge_media)=1)
; same => n,Dial(SIP/101)
; exten => 101,1,Set(_SIPSRTP_CRYPTO=enable)
exten => 101,1,Set(_SIP_SRTP_SDES=1)
exten => 101,2,Set(_SIPSRTP=enable)
exten => 101,3,Set(_SIPSRTP_CRYPTO=enable)
exten => 101,4,Dial(SIP/101)
exten => 600,1,NoOp( start)
exten => 600,n,NOOp( SECURE SIGNALING ${CHANNEL(secure_signaling)} )
exten => 600,n,NOOp( SECURE media ${CHANNEL(secure_media)} )
exten => 600,n,Answer()
exten => 600,n,Playback(demo-echotest)
exten => 600,n,Echo()
I think It could be a SRTP key sharing problem, how can I check if caller key is arrived to receiver phone ?
So all things are ok (I think) but I received in all environments the same problem “Noise and nothing at all” on SRTP/TLS call …
Someone had the same problem and has resolve it ? Or if I have wrong some config file could you help me
posting a sip.conf / extension.conf / X.conf that can help me to fix my problem ?
Could I check with some debug command (on CLI windows) if communication are encrypted or I have some config file problems …
Thanks for all help in advance…