Dial SIP URI of WebRTC endpoint

Asterisk Asterisk WebRTC
1

Hi,

I want to dial a SIP URI using PJSIP. The SIP URI is of the WebRTC endpoint.

I have created an endpoint with below configuration to dial out SIP URI. But, the outgoing INVITE is not going via asterisk’s http interface and rather it is going via standard UDP interface.

can someone help me understand what configuration I am missing to dial out a call on WebRTC endpoint.

[transport-ws]
type=transport
protocol=ws    
bind=0.0.0.0
external_signaling_address=x.x.x.x
external_media_address=x.x.x.x


[sipuri]
type=endpoint
transport=transport-ws
disallow=all
allow=ulaw,alaw
callerid=asterisk
context=outbound
webrtc=yes
direct_media=no
dtmf_mode=rfc4733
ice_support=yes
use_avpf=yes
dtls_verify=fingerprint
dtls_setup=actpass
media_use_received_transport=yes
rtcp_mux=yes
2

A WebRTC device has to be registered to Asterisk to contact it using a Websocket, in which case you would dial its endpoint or use the PJSIP_DIAL_CONTACTS dialplan function. Asterisk does not support creating a new outgoing Websocket connection.

3

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