Para master asterisk, saya ada kendala pengaturan asterisk webrtc dan webclient voip menggunakan JsSIP
untuk pengaturan webRTC saya sudah melihat di artikel ini Configuring Asterisk for WebRTC Clients - Asterisk Documentation
utk pengaturan jssip nya seperti yg ada di link berikut
tetapi setelah saya run tidak ada panggilan. berikut log di asterisk nya
== WebSocket connection from ‘192.168.43.17:52856’ for protocol ‘sip’ accepted using version ‘13’
– Added contact ‘sip:q717l0a5@192.168.43.17:52856;transport=ws;x-ast-orig-host=92j16gen600j.invalid:0’ to AOR ‘1000’ with expiration of 600 seconds
== Endpoint 1000 is now Reachable
[Oct 2 13:57:11] ERROR[2856]: iostream.c:647 ast_iostream_start_tls: Problem setting up ssl connection: error:00000001:lib(0)::reason(1), Internal SSL error
[Oct 2 13:57:11] ERROR[2856]: tcptls.c:179 handle_tcptls_connection: Unable to set up ssl connection with peer ‘192.168.43.17:52840’
[Oct 2 13:57:11] ERROR[2856]: iostream.c:552 ast_iostream_close: SSL_shutdown() failed: error:00000001:lib(0)::reason(1), Internal SSL error
[Oct 2 13:57:11] NOTICE[2828]: res_pjsip_session.c:4022 new_invite: 1000: Call (WSS:192.168.43.17:52856) to extension ‘6001’ rejected because extension not found in context ‘default’.
kira2 salah dimananya ya
dialcall sudah bisa dari web browser ke softphone. tp saat test audio/speaker tidak ada suara di mic atau di audio/speaker
berikut log webrtc
== WebSocket connection from ‘192.168.43.17:44760’ for protocol ‘sip’ accepted using version ‘13’
– Added contact ‘sip:s7v0q7n2@192.168.43.17:44760;transport=ws;x-ast-orig-host=9dhc1ptap738.invalid:0’ to AOR ‘1000’ with expiration of 600 seconds
== Endpoint 1000 is now Reachable
– Executing [6001@test:1] Answer(“PJSIP/1000-00000002”, “”) in new stack
> 0x7fdb38056020 – Strict RTP learning after remote address set to: 192.168.43.17:37199
> 0x7fdb38056020 – Strict RTP learning after ICE completion
> 0x7fdb38056020 – Strict RTP learning after remote address set to: 192.168.43.17:37199
> 0x7fdb38056020 – Strict RTP switching to RTP target address 192.168.43.17:37199 as source
– Executing [6001@test:2] Dial(“PJSIP/1000-00000002”, “PJSIP/6001,60”) in new stack
– Called PJSIP/6001
– PJSIP/1000-00000002 requested media update control 26, passing it to PJSIP/6001-00000003
– PJSIP/1000-00000002 requested media update control 26, passing it to PJSIP/6001-00000003
– PJSIP/6001-00000003 is ringing
== Everyone is busy/congested at this time (1:1/0/0)
– Executing [6001@test:3] Playback(“PJSIP/1000-00000002”, “vm-nobodyavail”) in new stack
– <PJSIP/1000-00000002> Playing ‘vm-nobodyavail.g722’ (language ‘en’)
> 0x7fdb38056020 – Strict RTP learning complete - Locking on source address 192.168.43.17:37199
– Auto fallthrough, channel ‘PJSIP/1000-00000002’ status is ‘BUSY’
apa ada yang kurang pengaturan di asterisk nya atau di sisi jssip nya
sudah lama tertidur website nya ya… 
mau jawab, tapi kawatir yg bertanya sudah entah kemana