บทความนี้แปลมาจาก https://www.vitalpbx.org/blog/asterisk-webrtc-from-scratch
เนื้อหาหลักๆ คือ ติดตั้ง asterisk 18 + PJSIP บน CentOS 7 คอนฟิก PJSIP ให้รองรับ WebRTC
คือ ใช้ web browser (chrome, firefox, Opera, etc.) ทำหน้าเป็นเครื่องโทรศัพท์ แทนที่ IP phone
หรือ softphone นั่นเอง
1. ติดตั้ง CentOS 7 minimal ตาม link
http://www.asterisk.in.th/voip/index.php/centos-linux/45-centos-7-minimal-install
2. ติดตั้ง software เพิ่มเติม
#yum install wget nano git epel-release
ยกเลิก SELINUX
#sed -i s/^SELINUX=.*$/SELINUX=disabled/ /etc/selinux/config
#reboot
3. ติดตั้ง Asterisk 18
#cd /usr/src
#wget http://downloads.astrerisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz
#tar -zxvf asterisk-18-current.tar.gz
#cd asterisk-18.*
#yum install svn
#./contrib/scripts/get_mp3_source.sh
#contrib/scripts/install_prereq install
#./configure --libdir=/urs/lib64 --with-jansson-bundled --with-pjproject-bundled
#make menuselect
อย่าลืมเลือก codec_opus ในหัวข้อ Codec Translators
#make && make install
#make samples
#make config
#groupadd asterisk
#useradd -r -d /var/lib/asterisk -g asterisk asterisk
#usermod -aG audio, dialout asterisk
#chown asterisk:asterisk -R /etc/asterisk
#chown asterisk:asterisk -R /var/{lib, log, spool}/asterisk
#chown -R asterisk:asterisk /usr/lib64/asterisk
#vi /etc/sysconfig/asterisk
AST_USER = "asterisk"
AST_GROUP = "asterisk"
#chkconfig asterisk on
#systemctl restart asterisk
#asterisk -rvvv
ถ้าได้ prompt asterisk ดังรูปก็ถือว่า OK
4. ติดตั้ง Apache
#yum install httpd
#systemctl enable httpd
#systemctl start httpd
#mkdir -p /var/www/html/ws.adventek.net/{public_html,logs}
*** ws.adventek.net คือ server ตัวที่เราติดตั้ง (อยู่บน cloud)
edit config file server ของเรา ดังนี้
#vi /etc/httpd/conf.d/ws.adventek.net.conf
NameVirtualHost *:80
<VirtualHost *:80>
ServerAdmin This email address is being protected from spambots. You need JavaScript enabled to view it.adventek.net
ServerName ws.adventek.net
ServerAlias ws.adventek.net
DocumentRoot /var/www/html/ws.adventek.net/public_html/
ErrorLog /var/www/html/ws.adventek.net/logs/error.log
CustomLog /var/www/html/ws.adventek.net/logs/access.log combined
</VirtualHost>
#system restart httpd
5. Create certificate
#yum install certbot python2-certbot-apache mod_ssl
#certbot --apache -d ws.adventek.net
แก้ไขไฟล์ /etc/httpd/conf.d/ssl.conf
|
#mv /etc/httpd/conf.d/ssl.conf /etc/httpd/conf.d/ssl.conf.bak
|
| #mv /etc/httpd/conf.d/welcome.conf /etc/httpd/conf.d/welcome.conf.bak #mv /etc/httpd/conf.d/ws.adventek.net.conf /etc/httpd/conf.d/ws.adventek.net.bak |
#systemctl restart httpd
#crontab -e
45 3 24 5,8,11,2 * /usr/bin/certbot renew && systemctl reload httpd
*** renew certificates ทุกๆ 3 เดือน ***
6. Config Asterisk
เช็คว่า Asterisk มี module เหล่านี้พร้อมหรือไม่
- res_crypto
- res_http_websocket
- res_pjsip_transport_websocket
- codec_opus
(ใช้คำสั่ง #asterisk -rx "module show")
#vi /etc/asterisk/http.conf
[general]
Servername = Asterisk
tlsbindaddr = 0.0.0.0:8089
bindaddr = 0.0.0.0
bindport = 8088
enabled = yes
tlsenable = yes
tlscertfile=/etc/letsencrypt/live/mydomian.com/fullchain.pem
tlsprivatekey=/etc/letsencrypt/live/mydomain.com/privkey.pem
แก้ไขไฟล์ /etc/asterisk/pjsip.conf ดังนี้
#mv /etc/asterisk/pjsip.conf /etc/asterisk/pjsip.conf.bak
#vi pjsip.conf
[system]
type=system
timer_t1=500
timer_b=32000
disable_tcp_switch=yes
[global]
type=global
max_initial_qualify_time=0
keep_alive_interval=90
contact_expiration_check_interval=30
default_voicemail_extension=*97
unidentified_request_count=3
unidentified_request_period=5
unidentified_request_prune_interval=30
mwi_tps_queue_high=500
mwi_tps_queue_low=-1
mwi_disable_initial_unsolicited=yes
send_contact_status_on_update_registration=yes
[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0:8089
allow_reload=yes
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
[webrtc-phones](!)
context=main-context
transport=transport-wss
allow=!all,opus,ulaw,alaw,vp8,vp9
webrtc=yes
[User1](webrtc-phones)
type=endpoint
callerid=”User One” <100>
auth=User1
aors=User1
[User1]
type=aor
max_contacts=3
[User1]
type=auth
auth_type=userpass
username=User1
password=1234
[User2](webrtc-phones)
type=endpoint
callerid=”User Two” <101>
auth=User2
aors=User2
[User2]
type=aor
max_contacts=3
[User2]
type=auth
auth_type=userpass
username=User2
password=1234
[User3](webrtc-phones)
type=endpoint
callerid=”User Three” <102>
auth=User3
aors=User3
[User3]
type=aor
max_contacts=3
[User3]
type=auth
auth_type=userpass
username=User3
password=1234
[User4]
type=endpoint
callerid="User Four" <714>
context=main-context
disallow=all
allow=ulaw
transport=transport-udp
auth=User4
aors=User4
[User4]
type=auth
auth_type=userpass
password=1234
username=User4
[User4]
type=aor
max_contacts=1
[User5]
type=endpoint
callerid="User Five" <715>
context=main-context
disallow=all
allow=ulaw
transport=transport-udp
auth=User5
aors=User5
[User5]
type=auth
auth_type=userpass
password=1234
username=User5
[User5]
type=aor
max_contacts=1
;;; end of file ;;;
User1 - User3 สำหรับ WebRTC
User4 - User5 สำหรับ IP phone หรือ softphone
เปลี่ยน file owner
#chown asterisk. /etc/asterisk/pjsip.conf
แก้ไขไฟล์ /etc/asterisk/extensions.conf
#mv /etc/asterisk/extensions.conf /etc/asterisk/extensions.conf.bak
#vi /etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=yes
priorityjumping=no
autofallthrough=no
[globals]
ATTENDED_TRANSFER_COMPLETE_SOUND=beep
[main-context]
include => from-extensions
include => subscriptions
include => textmessages
include => echo-test
include => speak-exte-nnum
[echo-test]
exten => 777,1,NoOp(FEATURE: ECHO TEST)
same => n,Answer
same => n,Wait(1)
same => n,Playback(demo-echotest)
same => n,Echo()
same => n,Playback(demo-echodone)
same => n,Hangup()
;END of [echo-test]
[textmessages]
exten => 100,1,Gosub(send-text,s,1,(User1))
exten => 101,1,Gosub(send-text,s,1,(User2))
exten => 102,1,Gosub(send-text,s,1,(User3))
exten => 714,1,Gosub(send-text,s,1,(User4))
exten => 715,2,Gosub(send-text,s,1,(User5))
[subscriptions]
exten => 100,hint,PJSIP/User1
exten => 101,hint,PJSIP/User2
exten => 102,hint,PJSIP/User3
exten => 714,hint,PJSIP/User4
exten => 715,hint,PJSIP/User5
[from-extensions]
; Feature Codes:
exten => *65,1,Gosub(moh,s,1)
; Extensions
exten => 100,1,Gosub(dial-extension,s,1,(User1))
exten => 101,1,Gosub(dial-extension,s,1,(User2))
exten => 102,1,Gosub(dial-extension,s,1,(User3))
exten => 714,1,Gosub(dial-extension,s,1,(User4))
exten => 715,1,Gosub(dial-extension,s,1,(User5))
exten => e,1,Hangup()
[moh]
exten => s,1,NoOp(Music On Hold)
exten => s,n,Ringing()
exten => s,n,Wait(2)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,MusicOnHold()
[dial-extension]
exten => s,1,NoOp(Calling: ${ARG1})
exten => s,n,Set(JITTERBUFFER(adaptive)=default)
exten => s,n,Dial(PJSIP/${ARG1},30)
exten => s,n,Hangup()
exten => e,1,Hangup()
[send-text]
exten => s,1,NoOp(Sending Text To: ${ARG1})
exten => s,n,Set(PEER=${CUT(CUT(CUT(MESSAGE(from),@,1),<,2),:,2)})
exten => s,n,Set(FROM=${SHELL(asterisk -rx ‘pjsip show endpoint ${PEER}’ | grep ‘callerid ‘ | cut -d’:’ -f2- | sed ‘s/^ *//’ | tr -d ‘‘)})
exten => s,n,Set(CALLERID_NUM=${CUT(CUT(FROM,>,1),<,2)})
exten => s,n,Set(FROM_SIP=${STRREPLACE(MESSAGE(from),
exten => s,n,MessageSend(pjsip:${ARG1},${FROM_SIP})
exten => s,n,Hangup()
;;; end of file ;;;
เปลี่ยน file owner
#chown asterisk. /etc/asterisk/extensions.conf
reload asterisk
| #asterisk -rvvv *CLI> module reload res_pjsip.so *CLI> dialplan reload |
7. ติดตั้ง WebRTC Client
#cd /var/www/html/ws.adventek.net/public_html
#git clone http://github.com/InnovateAsterisk/Browser-Phone.git
#chown -R apache:apache /var/www/html/ws.adventek.net/public_html/
8. ทดสอบ
ใช้ PC หรือ Notebook 2 เครื่อง เปิด web browser แล้ว connect ไปที่ ws.adventek.net แล้ว คอนฟิก account ดังนี้
- User3 (extension 102)
User1 (extension 100)
ลองโทรหากัน
