配置SRTP在星号1.8.1

我正在使用jitsi和星号1.8.1。我在他们之间成功配置了TLS。它的工作正常。现在我正在配置它们之间的SRTP。配置SRTP在星号1.8.1

以下是配置文件。

sip.conf

[general] 
context=incoming 
allowguest=no 
alwaysauthreject=yes 
allow=ulaw 
allow=alaw 
allow=gsm 


tlsenable=yes 
tlsbindaddr=0.0.0.0 
tlscertfile=/etc/asterisk/keys/newbie.pem 
tlscafile=/etc/asterisk/keys/ca.crt 
tlscipher=ALL 
tlsclientmethod=tlsv1 

[user1] 
type=peer 
defaultuser=user1 
secret=1000 
dtmfmode=rfc2833 
callerid="User one" 
host=dynamic  ; The device must always register 
canreinvite=no 
nat=yes 
encryption=yes 
transport=tls 

; Deny registration from anywhere first 
deny=0.0.0.0/0.0.0.0 
; Replace the IP address and mask below with the actual IP address and mask 
; of the computer running the softphone, or the address of the hardware phone, 
; either a host address and full mask, or a network address and correct mask, 
; registering will be allowed from that host/network. 
permit=192.168.51.0/255.255.255.0 
context=myphones 

[user2] 
type=peer 
defaultuser=user2 
secret=1001 
dtmfmode=rfc2833 
callerid="User two" 
host=dynamic  ; The device must always register 
canreinvite=no 
nat=yes 
encryption=yes 
transport=tls 

; Deny registration from anywhere first 
deny=0.0.0.0/0.0.0.0 
; Replace the IP address and mask below with the actual IP address and mask 
; of the computer running the softphone, or the address of the hardware phone, 
; either a host address and full mask, or a network address and correct mask, 
; registering will be allowed from that host/network. 
permit=192.168.51.0/255.255.255.0 
context=myphones

extension.conf

[general] 
static=yes 
writeprotect=no 
clearglobalvars=no 

[incoming] 
exten => s,1,Hangup() 

[myphones] 
exten => user1,1,Set(CHANNEL(secure_bridge_signaling)=1) 
exten => user1,n,Set(CHANNEL(secure_bridge_media)=1) 
exten => user1,n,Dial(SIP/user1) 
exten => user1,n,Hangup() 

exten => user2,1,Set(CHANNEL(secure_bridge_signaling)=1) 
exten => user2,n,Set(CHANNEL(secure_bridge_media)=1) 
exten => user2,n,Dial(SIP/user2) 
exten => user2,n,Hangup() 

exten => 201,1,Answer() 
exten => 201,n,Playback(tt-monty-knights) 
exten => 201,n,Hangup() 
exten => 202,1,Answer() 
exten => 202,n,Playback(welcome) 
exten => 202,n,Playback(demo-echotest) 
exten => 202,n,Echo() 
exten => 202,n,Playback(demo-echodone) 
exten => 202,n,Playback(vm-goodbye) 
exten => 202,n,Hangup()

我上传SRTP模块还。它被加载。但是，当user1呼叫user2我的星号服务器出现分段故障并关闭。

这里有任何帮助，如何配置srtp。我是否做了正确的或任何的问题是有..

我已经提前在http://forums.asterisk.org/viewtopic.php?f=1&t=84587

感谢张贴这个问题，

来源

2012-10-25 Newbie123

你奶源来解决问题？如果是PLZ股 – dakait

Asterisk的1.8有SRTP的原生支持。所以你可以用Asterisk 1.8.x服务器轻松地尝试。普通的SIP电话不支持这个系统。您需要查找SRTP支持和TLS支持的IP /软电话进行设置。使用blink软电话（http://www.icanblink.com）。我遵循以下文章来设置安全的VoIP系统。

对于SRTP：http://www.remiphilippe.fr/2011/01/16/asterisk-srtp-with-1-8/

对于TLS：https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial，（你可以从这里找到ast_tls_cert脚本：https://reviewboard.asterisk.org/r/979/diff/）

来源

2012-10-25 05:07:59

我已经通过此链接了。 Mine TLS在jitsi和星号之间正常工作。但是当我启用SRTP双方我的星号服务器突然关闭。 – Newbie123

我不确定这里发生了什么，但上面的步骤对我有用。尝试升级你的星号。 –