Asterisk 1.8.1中SRTP的配置

Question

我正在使用jitsi和asterisk 1.8.1。我成功地在它们之间配置了TLS。它工作得很好。现在，我正在配置它们之间的SRTP。

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()

我也上传srtp模块。上膛了。但是当user1调用user2时，我的asterisk服务器出现分段故障并关闭。

这里有关于如何配置srtp的帮助吗？我是做对了还是有什么问题..

我已经在http://forums.asterisk.org/viewtopic.php?f=1&t=84587上发布了这个问题

提前THanks。，

Answer 1

Asterisk 1.8具有对SRTP的本地支持。因此，您可以很容易地在Asterisk 1.8.x服务器上尝试此操作。此系统不支持普通SIP电话。您需要找到支持SRTP和TLS的IP/软件电话进行设置。为此，请使用闪烁软电话(http://www.icanblink.com)。我按照下面的文章来设置安全的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 (您可以从以下位置找到ast_tls_cert脚本：https://reviewboard.asterisk.org/r/979/diff/)

Answer 2

我知道这是一个老问题，但是我也有同样的问题。我理解了这个解释，所以我想我应该把它放在最后，以防其他人遇到同样的问题(或者在我们的例子中是缺乏理解！)

帖子可以在这里找到：http://permalink.gmane.org/gmane.comp.voip.sip-communicator.user/921

ZRTP确实支持

，但它是通过ZRTP密钥协商来实现的。

然而，Asterisk不支持它，因此您需要使用实际的SIP和RTP代理才能使其工作。

希望这能帮上忙

埃米尔·