Websocket連接失敗並顯示星號11

Question

我正在嘗試將websocket配置爲使用星號11.但存在一些問題。 我遵循的步驟是： 在啓用的http.conf以下

enabled=yes 
bindaddr=0.0.0.0 
bindport=8088 

我也配置了DTLS支持星號。 但是，當我嘗試連接到websocket新的WebSocket（「ws：// mySeverIp：8088/ws」）; 。它會引發錯誤

WebSocket connection failed: Error during WebSocket handshake: Unexpected response code: 400 

任何人都請幫忙。

謝謝

Answer 1

這是一個完整的安裝指南。請讓我知道這是否解決了您的問題。 星號還提供了wiki post on the matter

安裝SRTP：

cd ~ 
git clone https://github.com/cisco/libsrtp.git 
cd libsrtp/ 
autoconf 
./configure CFLAGS=-fPIC --prefix=/usr 
make 
make runtest 
sudo make install 

安裝PJPROJECT：

cd ~ 
git clone https://github.com/asterisk/pjproject pjproject 
cd pjproject/ 
./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr --with-external-srtp 
make 
sudo make install 

安裝UUID開發庫（不需要爲Asterisk 12）：

cd ~ 
sudo apt-get install uuid-dev -y (for Debian & ubuntu, libuuid-devel for CentOS) 

安裝星號11：

cd ~ 
wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11-current.tar.gz 
tar -xzf asterisk-11-current.tar.gz 

生成和SRTP安裝DTLS證書：

cd asterisk*/contrib/scripts 
sudo mkdir /etc/asterisk/ 
sudo mkdir /etc/asterisk/keys/ 

要生成自簽名的SSL證書，請使用以下命令：

### Replace 10.x.x.x by the IP adress of your server. 10.x.x.x is intern, use a public IP if your Asterisk will be exposed over Internet. 
sudo ./ast_tls_cert -C 10.x.x.x -O "Your Company" -d /etc/asterisk/keys 

在這個過程中，你會被要求選擇一個關鍵。按下Enter鍵，每次輸入相同的密鑰並生效。

如果將證書頒發機構生成你的SSL證書，請使用以下方法：
http://codeghar.wordpress.com/2013/04/16/generate-certificate-signing-request-on-linux/ http://codeghar.wordpress.com/2013/04/16/use-private-certificate-authority-to-sign-certificate-signing-request-on-linux/

在這個例子中的證書路徑是/etc/asterisk/keys/asterisk.pem

安裝星號（是的，你需要使用PJPROJECT和LIBSRTP編譯星號）：

cd ~ 
cd asterisk* 
sudo ./configure --with-pjproject --with-ssl --with-srtp 
make menuselect 

檢查軟件包pbx_realtime,res_odbc,res_http_websocket,res_crypto和chan_sip被激活。這是在Asterisk中通過WS或WSS使用WebRTC的必備條件。

make 
sudo make install 
sudo make config 
## Recommended demo conf files with : 
sudo make samples 
cd ~ 

激活在/etc/asterisk/http.conf的WebSockets ANS SecureWebSockets（文件，該文件管理的Apache HTTP網頁Asterisk的實例）。如果使用Asterisk Realtime（ODBC），則必須在每個對等項中指定文件（行dtlscertfile & dtlsprivatekey表sippeers）。 ：

enabled=yes; 
bindport=8088; 
tlscertfile=/etc/asterisk/keys/asterisk.pem 
tlsprivatekey=/etc/asterisk/keys/asterisk.pem 

確保對文件夾權限都不錯：

sudo chown AsteriskUser. /var/run/asterisk 
sudo chown -R AsteriskUser. /etc/asterisk 
sudo chown -R AsteriskUser. /var/{lib,log,spool}/asterisk 
sudo chown -R AsteriskUser. /usr/lib/asterisk 

（通過運行Asterisk的服務用戶更換AsteriskUser）在sip.conf創建您的WebRTC同行（複製再拍用戶）：

[1060] ; This will be WebRTC client 
type=friend ; 
username=1060 ; The Auth user for SIP.js 
host=dynamic ; Allows any host to register 
secret=password ; The SIP Password for SIP.js 
encryption=yes ; Tell Asterisk to use encryption for this peer 
avpf=yes ; Tell Asterisk to use AVPF for this peer 
icesupport=yes ; Tell Asterisk to use ICE for this peer 
context=default ; Tell Asterisk which context to use when this peer is dialing 
directmedia=no ; Asterisk will relay media for this peer 
transport=udp,ws,wss ; Asterisk will allow this peer to register on UDP or WebSockets 
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11 
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer 
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs 
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is 
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is 
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS 

[1061] ; This will be the legacy SIP client 
type=friend; 
username=1061; 
host=dynamic; 
secret=password; 
context=default; 

編輯的extensions.conf讓每一個同行呼籲：

[default] 
exten => 1060,1,Dial(SIP/1060) ; Dialing 1060 will call the SIP client registered to 1060 
exten => 1061,1,Dial(SIP/1061) ; Dialing 1061 will call the SIP client registered to 1061 

啓動Asterisk的服務：

sudo service asterisk start 

打開所需的端口：

[Ubuntu] : 
sudo ufw allow 5060 (or whatever port you have choosen in sip.conf `udpbindaddr=10.x.x.x:5060; tcpbindaddr=10.x.x.x:5060;tlsbindaddr=10.x.x.x:5061`) 
sudo ufw allow 5061 
sudo ufw allow 8088 (or whatever port you have choosen in http.conf : `bindport=8088`) 
sudo ufw allow 8089 (or whatever port you have choosen in http.conf : `tlsbindaddr=10.x.x.x:8089`) 
sudo ufw allow 10000:20000/udp (or whatever range you have choosen in rtp.conf : `rtpstart=10000; rtpend=20000`) 
[or if you are on Debian] : 
sudo iptables -A INPUT -p tcp --dport 5060 -j ACCEPT 
sudo iptables -A INPUT -p tcp --dport 5061 -j ACCEPT 
sudo iptables -A INPUT -p tcp --dport 8088 -j ACCEPT 
sudo iptables -A INPUT -p tcp --dport 8089 -j ACCEPT 
sudo iptables -A INPUT -p tcp --match multiport --dports 10000:20000 -j ACCEPT 

重啓（或啓動）的服務：從另一臺機器sudo service asterisk restart

測試的WebSockets：與sudo apt-get install wscat –y

安裝WSCAT

## If error "connect ECONNREFUSED" it's not OK. 
## If Echo service returns your messages, it's OK. 
wscat -s echo -c ws://10.x.x.x:8088/ws 
## The same command with WSS should work if you've installed WSS. 

使用JavaScript客戶端（例如JsSIP，sipML5，WebRTComm）測試您的SIP over WebSocket ...

使用sudo asterisk -vvvvvv -g -dddddd -r訪問SIP控制檯進行調試和跟蹤。

要做與Asterisk 12相同的工作，只需在Asterisk安裝中用Asterisk-12替換Asterisk-11即可。

在這裏，您可以使用Realtime，WS，WSS（省略ODBC conf）找到完整的Asterisk 12 conf文件。我張貼，因爲你可能會發現有用的檢查，如果一些參數在缺少你安裝：

的http.conf

; 
; Asterisk Builtin mini-HTTP server 
; 
[general] 
enabled=yes; 
bindaddr=10.x.x.x; 
bindport=8088; 
tlsenable=yes   ; enable tls - default no. 
tlsbindaddr=10.x.x.x:8089 ; address and port to bind to - default is bindaddr and port 8089. 
tlscertfile=/etc/asterisk/keys/asterisk.pem ; path to the certificate file (*.pem) only. 
tlsprivatekey=/etc/asterisk/keys/asterisk.pem ; path to private key file (*.pem) only. 

extensions.conf文件（用於實時做出!!）

[general] 
[globals] 
; 
[default] 
switch =>Realtime 

modules.conf（用於Realtime !!!）

[modules] 
autoload=yes 
preload => res_odbc.so 
preload => res_config_odbc.so 
noload => pbx_gtkconsole.so 
load => res_musiconhold.so 
noload => chan_alsa.so 
noload => chan_console.so 

extconfig.conf（用於實時!!!）

[settings] 
sippeers => odbc,YourAsteriskrealtimeDB,sippeers 
sipusers => odbc,YourAsteriskrealtimeDB,sippeers 
extensions => odbc,YourAsteriskrealtimeDB,extensions 
ps_endpoints => odbc,YourAsteriskrealtimeDB,ps_endpoints 
ps_auths => odbc,YourAsteriskrealtimeDB,ps_auths 

的asterisk.conf

[directories](!) 
astetcdir => /etc/asterisk 
astmoddir => /usr/lib/asterisk/modules 
astvarlibdir => /var/lib/asterisk 
astdbdir => /var/lib/asterisk 
astkeydir => /var/lib/asterisk 
astdatadir => /var/lib/asterisk 
astagidir => /var/lib/asterisk/agi-bin 
astspooldir => /var/spool/asterisk 
astrundir => /var/run/asterisk 
astlogdir => /var/log/asterisk 
astsbindir => /usr/sbin 
[options]; 
verbose = 3; 
debug = 3; 
highpriority = yes  ; Run realtime priority (same as -p at startup). 
initcrypto = yes  ; Initialize crypto keys (same as -i at startup). 
[compat]; 
pbx_realtime=1.6; 
res_agi=1.6; 
app_set=1.6; 

rtp.conf

; 
; RTP Configuration 
; 
[general]; 
rtpstart=10000; 
rtpend=20000; 
icesupport=true; 
stunaddr=stun.l.google.com:19302; 

sip.conf

; 
; SIP Configuration for Asterisk 
; 
[general] 
context=default ; Default context for incoming calls. Defaults to 'default' 
allowoverlap=no     ; Disable overlap dialing support. (Default is yes) 
realm=YourAsteriskREALM    ; Realm for digest authentication 
           ; defaults to "asterisk". If you set a system name in 
           ; asterisk.conf, it defaults to that system name 
           ; Realms MUST be globally unique according to RFC 3261 
           ; Set this to your host name or domain name 

udpbindaddr=10.x.x.x    ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) 
           ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) 
videosupport=yes    ; Turn on support for SIP video. You need to turn this 
           ; on in this section to get any video support at all. 
           ; You can turn it off on a per peer basis if the general 
           ; video support is enabled, but you can't enable it for 
           ; one peer only without enabling in the general section. 
           ; If you set videosupport to "always", then RTP ports will 
           ; always be set up for video, even on clients that don't 
           ; support it. This assists callfile-derived calls and 
           ; certain transferred calls to use always use video when 
           ; available. [yes|NO|always] 
rtsavepath=yes     ; If using dynamic realtime, store the path headers 
send_diversion=no    ; Default "yes"  ; Asterisk normally sends Diversion headers with certain SIP 
                ; invites to relay data about forwarded calls. If this option 
                ; is disabled, Asterisk won't send Diversion headers unless 
                ; they are added manually. 
rtpkeepalive=2   ; Send keepalives in the RTP stream to keep NAT open (default is off - zero)(secs) 
;--------------------------- SIP DEBUGGING --------------------------------------------------- 
sipdebug = yes     ; Turn on SIP debugging by default, from 
           ; the moment the channel loads this configuration 
icesupport = yes; 
;----------------------------------------- REALTIME SUPPORT ------------------------ 
; For additional information on ARA, the Asterisk Realtime Architecture, 
; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration 
; 
rtcachefriends=yes    ; Cache realtime friends by adding them to the internal list 
           ; just like friends added from the config file only on a 
           ; as-needed basis? (yes|no) 

rtsavesysname=yes    ; Save systemname in realtime database at registration 
           ; Default= no 

rtupdate=yes     ; Send registry updates to database using realtime? (yes|no) 
           ; If set to yes, when a SIP UA registers successfully, the ip address, 
           ; the origination port, the registration period, and the username of 
           ; the UA will be set to database via realtime. 
           ; If not present, defaults to 'yes'. Note: realtime peers will 
           ; probably not function across reloads in the way that you expect, if 
           ; you turn this option off. 
rtautoclear=yes    ; Auto-Expire friends created on the fly on the same schedule 
           ; as if it had just registered? (yes|no|<seconds>) 
           ; If set to yes, when the registration expires, the friend will 
           ; vanish from the configuration until requested again. If set 
           ; to an integer, friends expire within this number of seconds 
           ; instead of the registration interval. 

[authentication] 
; 
; DTLS-SRTP CONFIGURATION 
; 
dtlsenable = yes     ; Enable or disable DTLS-SRTP support 
dtlsverify = no     ; Verify that provided peer certificate and fingerprint are valid 
dtlscertfile=/etc/asterisk/keys/asterisk.pem    ; Path to certificate file to present 
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem    ; Path to private key for certificate file 
dtlssetup = actpass    ; Whether we are willing to accept connections, connect to the other party, or both. 

[basic-options](!)    ; a template 
     dtmfmode=rfc2833 
     context=from-office 
     type=friend 

[natted-phone](!,basic-options) ; another template inheriting basic-options 
     directmedia=no 
     host=dynamic 

[public-phone](!,basic-options) ; another template inheriting basic-options 
     directmedia=yes 

[my-codecs](!)     ; a template for my preferred codecs 
     disallow=all 
     allow=ilbc 
     allow=g729 
     allow=gsm 
     allow=g723 
     allow=ulaw 

[ulaw-phone](!)     ; and another one for ulaw-only 
     disallow=all 
     allow=ulaw 

[1060] ; This will be WebRTC client 
type=friend 
username=1060 ; The Auth user for SIP.js 
host=dynamic ; Allows any host to register 
secret=password ; The SIP Password for SIP.js 
encryption=yes ; Tell Asterisk to use encryption for this peer 
avpf=yes ; Tell Asterisk to use AVPF for this peer 
icesupport=yes ; Tell Asterisk to use ICE for this peer 
context=default ; Tell Asterisk which context to use when this peer is dialing 
directmedia=no ; Asterisk will relay media for this peer 
transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets 
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11 
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer 
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs 
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is 
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is 
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS 

[1061] ; This will be the legacy SIP client 
type=friend 
username=1061 
host=dynamic 
secret=password 
context=default 

res_odbc.conf（只爲Asterisk實時!!!!）

[YourAsteriskrealtimeDB] 
enabled => yes 
dsn => your-asterisk-BD-connector-name-as-defined-in-file-odbcinst.ini 
username => YourMySQLUser 
password => YourMySQLPassword 
pre-connect => yes 

如果使用實時，插入下面的通用撥號方案：

INSERT INTO `extensions` ( `context`, `exten`, `priority`,  `app`, `appdata` ) VALUES ( 'default', '_X.', 1, 'Dial',  'SIP/${EXTEN}' ); 

Answer 2

添加證書細節的http.conf。以下爲我工作。

tlsenable=yes 
tlsbindaddr=0.0.0.0:8089 
tlscertfile=/path-to/cert.pem 
tlsprivatekey=/path-to/privkey.pem