Asterisk 11.17.0
JsSIP 0.6.21
Chrome 34.0.1847.137
sip.conf
avpf=yes
force_avp=yes
encryption=yes
icesupport=yes
dtlsverify=no
dtlscertfil=/etc/asterisk/ssl/asterisk.pem
dtlsprivatekey=/etc/asterisk/ssl/asterisk.pem
dtlssetup=actpass
extensions.conf
exten => 01231,1,Noop(Test EA)
same => n,Ringing()
same => n,Wait(4)
same => n,Progress
same => n,Playback(/var/lib/asterisk/moh/default/macroform-cold_day,noanswer)
same => n,Hangup
Make call to 01231 , this logs after Progress()
With “dtlsenable=yes” not playing music in webphone.
[code]<— Transmitting (no NAT) to 192.168.200.157:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/WSS 7lqnkt03oon5.invalid;branch=z9hG4bK4328316;received=192.168.200.157
From: “1999” sip:1999@192.168.100.203;tag=tdri99671e
To: sip:01231@192.168.100.203;tag=as58b79374
Call-ID: d35803n8uakpu0d21q9p
CSeq: 1827 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:01231@192.168.100.203:5060;transport=WS
Content-Type: application/sdp
Content-Length: 622
v=0
o=Asterisk 981451849 981451849 IN IP4 192.168.100.203
s=SIP Call
c=IN IP4 192.168.100.203
t=0 0
m=audio 16942 RTP/SAVPF 8 126
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=ice-ufrag:693c3fec50714d095060243c5b04edeb
a=ice-pwd:7a5e760e12eb56dd00204e2e06b5420f
a=candidate:Hc0a864cb 1 UDP 2130706431 192.168.100.203 16942 typ host
a=candidate:Hc0a864cb 2 UDP 2130706430 192.168.100.203 16943 typ host
a=connection:new
a=setup:active
a=fingerprint:SHA-256 C6:C4:EA:08:C3:AB:D9:D4:CD:04:EB:2A:05:08:DF:C4:1E:39:A1:1D:C6:59:A9:CB:ED:35:13:EE:3D:64:F8:35
a=sendrecv
[/code]
With “dtlsenable=no” playing music in webphone.
[code]<— Transmitting (no NAT) to 192.168.200.157:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/WSS 7lqnkt03oon5.invalid;branch=z9hG4bK6363661;received=192.168.200.157
From: “1999” sip:1999@192.168.100.203;tag=51b4ncn2rq
To: sip:01231@192.168.100.203;tag=as4acd29c9
Call-ID: d35801m50nn41hed2ff8
CSeq: 7916 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:01231@192.168.100.203:5060;transport=WS
Content-Type: application/sdp
Content-Length: 553
v=0
o=Asterisk 397923233 397923233 IN IP4 192.168.100.203
s=SIP Call
c=IN IP4 192.168.100.203
t=0 0
m=audio 16922 RTP/SAVPF 8 126
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=ice-ufrag:08d41c9c0da24678770a950a0990a7a0
a=ice-pwd:6426df7c07422812487fb1fd5f50ec81
a=candidate:Hc0a864cb 1 UDP 2130706431 192.168.100.203 16922 typ host
a=candidate:Hc0a864cb 2 UDP 2130706430 192.168.100.203 16923 typ host
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:BR8CLbJ0tbGK6XGMbHsoB9chUB34xl5f5qz/axk/
[/code]
Why ? Problem with Asterisk or JsSIP or other ?