I have setup a asterisk on openwrt. everything works fine except calls between webrtc extension and sip extension. Asterisk version is 18
here is the output of httpstatus
http.conf is as follows
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
;tlscertfile=/etc/asterisk/keys/fullchain.pem
;tlsprivatekey=/etc/asterisk/keys/privkey.pem
pjsip.conf is as follows
;=========== General settings ===========
[global]
type=global
user_agent=Asterisk PBX
debug=yes
;externip = 192.168.1.6
;localnet=172.17.0.0/255.255.0.0
nat=yes
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
[transport-tcp]
type=transport
protocol=tcp
bind=0.0.0.0
[transport-ws]
type=transport
protocol=ws
bind=0.0.0.0
;=========== Extension 300 ===========
[300]
type=endpoint
context=from-internal
disallow=all
allow=alaw,ulaw
allow=h264,vp8
auth=300
aors=300
callerid=300
identify_by=username,auth_username
;media_address=192.168.1.6
[300]
type=auth
auth_type=userpass
password=300
username=300
[300]
type=aor
max_contacts=1
[400]
type=aor
max_contacts=5
remove_existing=yes
[400]
type=auth
auth_type=userpass
username=400
password=400
[400]
type=endpoint
aors=400
auth=400
callerid=400
dtls_auto_generate_cert=yes
webrtc=yes
; Setting webrtc=yes is a shortcut for setting the following options:
use_avpf=yes
; media_encryption=dtls
; dtls_verify=fingerprint
; dtls_setup=actpass
; ice_support=yes
media_use_received_transport=yes
rtcp_mux=yes
context=from-internal
disallow=all
allow=alaw,ulaw
allow=h264,vp8
identify_by=username,auth_username
[500]
type=aor
max_contacts=5
remove_existing=yes
[500]
type=auth
auth_type=userpass
username=500
password=500
[500]
type=endpoint
aors=500
auth=500
callerid=500
dtls_auto_generate_cert=yes
webrtc=yes
; Setting webrtc=yes is a shortcut for setting the following options:
use_avpf=yes
; media_encryption=dtls
; dtls_verify=fingerprint
; dtls_setup=actpass
; ice_support=yes
media_use_received_transport=yes
rtcp_mux=yes
context=from-internal
disallow=all
allow=alaw,ulaw
allow=h264,vp8
identify_by=username,auth_username
WebSocket connection from '192.168.1.182:37946' for protocol 'sip' accepted using version '13'
-- Added contact 'sip:g6410242@192.168.1.182:37946;transport=ws' to AOR '500' with expiration of 600 seconds
== Endpoint 500 is now Reachable
-- Executing [300@fullrights:1] Dial("PJSIP/500-00000000", "PJSIP/300/sip:300@192.168.1.184:17824") in new stack
-- Called PJSIP/300/sip:300@192.168.1.184:17824
-- PJSIP/300-00000001 is ringing
> 0x155d730 -- Strict RTP learning after remote address set to: 192.168.1.184:45198
> 0x1562740 -- Strict RTP learning after remote address set to: 192.168.1.184:21364
-- PJSIP/300-00000001 answered PJSIP/500-00000000
> 0x158b970 -- Strict RTP learning after remote address set to: 5.36.210.24:45718
> 0x157fbe0 -- Strict RTP learning after remote address set to: 5.36.210.24:45718
-- Channel PJSIP/300-00000001 joined 'simple_bridge' basic-bridge <7882e14c-909e-42c4-91f6-e07dc77a892d>
-- Channel PJSIP/500-00000000 joined 'simple_bridge' basic-bridge <7882e14c-909e-42c4-91f6-e07dc77a892d>
> 0x158b970 -- Strict RTP learning after remote address set to: 192.168.1.182:45718
> 0x157fbe0 -- Strict RTP learning after ICE completion
> 0x158b970 -- Strict RTP learning after remote address set to: 192.168.1.182:45718
> 0x157fbe0 -- Strict RTP learning after remote address set to: 192.168.1.182:45718
> 0x155d730 -- Strict RTP switching to RTP target address 192.168.1.184:45198 as source
> 0x1562740 -- Strict RTP switching to RTP target address 192.168.1.184:21364 as source
> 0x155d730 -- Strict RTP learning complete - Locking on source address 192.168.1.184:45198
> 0x1562740 -- Strict RTP learning complete - Locking on source address 192.168.1.184:21364
-- Channel PJSIP/300-00000001 left 'simple_bridge' basic-bridge <7882e14c-909e-42c4-91f6-e07dc77a892d>
-- Channel PJSIP/500-00000000 left 'simple_bridge' basic-bridge <7882e14c-909e-42c4-91f6-e07dc77a892d>
== Spawn extension (fullrights, 300, 1) exited non-zero on 'PJSIP/500-00000000'
-- Executing [300@fullrights:1] Dial("PJSIP/500-00000002", "PJSIP/300/sip:300@192.168.1.184:17824") in new stack
-- Called PJSIP/300/sip:300@192.168.1.184:17824
-- PJSIP/300-00000003 is ringing
> 0x13fcf10 -- Strict RTP learning after remote address set to: 192.168.1.184:22554
-- PJSIP/300-00000003 answered PJSIP/500-00000002
> 0x155d520 -- Strict RTP learning after remote address set to: 5.36.210.24:50974
-- Channel PJSIP/300-00000003 joined 'simple_bridge' basic-bridge <10012695-3d60-498c-837d-e513f6299e6f>
-- Channel PJSIP/500-00000002 joined 'simple_bridge' basic-bridge <10012695-3d60-498c-837d-e513f6299e6f>
> 0x155d520 -- Strict RTP learning after ICE completion
> 0x155d520 -- Strict RTP learning after remote address set to: 192.168.1.182:50974
> 0x13fcf10 -- Strict RTP switching to RTP target address 192.168.1.184:22554 as source
> 0x13fcf10 -- Strict RTP learning complete - Locking on source address 192.168.1.184:22554
-- Channel PJSIP/300-00000003 left 'simple_bridge' basic-bridge <10012695-3d60-498c-837d-e513f6299e6f>
-- Channel PJSIP/500-00000002 left 'simple_bridge' basic-bridge <10012695-3d60-498c-837d-e513f6299e6f>
== Spawn extension (fullrights, 300, 1) exited non-zero on 'PJSIP/500-00000002'
The call arrives and when I pick it up there is no audio and no video .
When I dial from webrtc to webrtc client 400-500 even then it doesnt work properly, No audi video
UPDATE It seems that the dtls failure is the reason for failed audio video but why is it failing ? How can that be stopeed?