PJSIP and webrtc from AWS

Asterisk Asterisk WebRTC
1

Hi - I am trying to get webrtc running on AWS. I have used webrtc locally and works fine.

I have All the ports open on the AWS firewall. A linphone client works just fine with audio on the AWS server. I have added the external_media_address and external_signaling_address to pjsip.conf and set the local_net.

My setup for webrtc is: (I am using sipml5 as the client)

[webrtc1]
type=wizard
accepts_registrations=yes
sends_registrations=no
accepts_auth=yes
sends_auth=no
inbound_auth/username=webrtc1
inbound_auth/password=set
aor/max_contacts=1
aor/remove_existing=yes
endpoint/rtp_symmetric=yes
endpoint/force_rport=yes
endpoint/direct_media=no
endpoint/context=context-webrtc1
aor/qualify_frequency=180
aor/remove_existing=yes
endpoint/allow=ulaw,alaw,gsm,opus,vp8,h264
endpoint/webrtc=yes
endpoint/dtls_cert_file=/etc/asterisk/certs/fullchain.pem
endpoint/dtls_private_key=/etc/asterisk/certs/privkey.pem

I dont get audio or video on my browser client.

What might I be missing. The softphone works, and webrtc works on a local server.

Thanks

Jerry

2

WebRTC uses ICE, which has candidates and these are configured separately:

asterisk/configs/samples/rtp.conf.sample at master · asterisk/asterisk

3

I have used webrtc “locally” for some time - without STUN or ICE in play. I am just trying to move the setup to AWS. We have all the certificates (LetsEncypt) on the server and for asterisk. A softphone works with AWS - but the webrtc I get nothing …. I would rather not use ICE or anything additional. Why/how can I take what works locally and get to work on AWS ?

Thanks

Jerry

4

You’ve used STUN and ICE, it is fundamental technology used by WebRTC. There is no “not using it”. The browser requires it, just like it requires DTLS-SRTP. Even local, it would have been used.

And to go into more detail. Locally it would have ICE candidates that are, well, local IP address and port. Each side would exchange, figure out that it works, and tada - you’ve got media flow. With Asterisk being remote then the best option to ensure that media flow occurs unhindered is to configure rtp.conf so it places the public IP address as an ICE candidate, and ensuring the RTP ports are forwarded.

5

Thanks you jcolp.

yes - I added in rtp.conf

[ice_host_candidates]

; => ,[include_local_address]

172.X.X.X => Y.Y.Y.Y

and this works. Curious why that just happen automatically with external_media_address set.

Either way - THANKS

Jerry