问题描述
我正在尝试使用星号ans CSipSimple作为客户端来设置voip交换，以下是详细信息
Im trying to setup voip exchange using asterisk ans CSipSimple as client, fol are the detials
服务器端:
- 为服务器和两个客户端生成证书
- 将服务器证书放置在/etc/asterisk/keys/ 中
sip.conf:
[general]
context=local
allowguest=no
alwaysauthreject=yes
allow=gsm
allow=ulaw
allow=alaw
directmedia=yes
allowoverlap=no
bindport=5061
tlsdontverifyserver=yes
tlsenable=yes
tlsbindaddr=192.168.0.119
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscapath=/etc/asterisk/keys
register => tls://john:[email protected]:5061
register => tls://jane:[email protected]:5061
tlscipher=ALL
tlsclientmethod=tlsv1
localnet=192.168.0.119/255.255.255.0
[john]
type=peer
defaultuser=john
secret=password
dtmfmode=rfc2833
callerid="User one"
host=dynamic
canreinvite=no
nat=no
encryption=yes
transport=tls
[jane]
type=peer
defaultuser=jane
secret=password
dtmfmode=rfc2833
callerid="User two"
host=dynamic
canreinvite=no
nat=no
encryption=yes
transport=tls
客户端:
在OSTN帐户中设置TLS设置
setup the TLS setting in OSTN account
问题
显示TLS + zRTP并拨打电话而是在星号CLI上看到一个消息，
TLS +zRTP displayed and the call is madeinstead on the asterisk CLI i see a messsage,
我使用android 4.2.2星号1.8，如果我将srtp强制设置为zRTP =>创建zrtp，则会进行呼叫，指示立即跳+ srtp的TLS
im using android 4.2.2 asterisk 1.8, if i make srtp mandatory and zRTP => create zrtp the call is made indicating TLS to the immediate hop + srtp
如何使用CSipSimple作为客户端在星号上实现TLS + ZRTP.
How can i achieve TLS +ZRTP on asterisk using CSipSimple as client.
致谢.
推荐答案
ZRTP是一种用于端到端设备加密的协议，而使用标准的未经修改的Asterisk则无法实现，因为它基本上是作为服务器而不是作为服务器来设计的.代理(这是实现zrtp转发所需要的).但是，Zfgone项目为Asterisk发布了一些补丁程序来支持它.由于星号上的一些错误，由于在旅途中可能会发生一些问题(例如，某些编解码器引起的声音失真)，因此实现起来也是一个挑战，并且这还意味着某些功能限制(例如，无法转接或保持通话) ).但是您可以使用SRTP(它也加密流量，但仅加密服务器和客户端之间的流量)如果要使用zrtp，我建议您使用kamailio sip服务器.
ZRTP is a protocol for end-to-end devices encryption and this cannot be achieved with the standard unmodified Asterisk since it is basically designed as a server not as a proxy (which is what you need in order to achieve zrtp forwarding). However, The Zfgone project has released some patches for Asterisk to support it. The implementation is quite a challenge since some problems may occurs on the go (for example, sound distortion caused by some codecs) due some bugs on asterisk and it also implies some feature limitations (like not being able to transfer or put a call on hold). But you can use SRTP (it also encrypts traffic but only between server and client)If you want to use zrtp, I will recommend you to use a kamailio sip server.
这篇关于使用星号和CSipSimple为VOIP设置TLS + ZRTP的文章就介绍到这了，希望我们推荐的答案对大家有所帮助，也希望大家多多支持！