Hello Marek! I’ve been running on an issue with my Asterisk 12 configuration
for using WebRTC on a LAN environment for about a month! I really need some
help …
My calls from the browser are done fine. I get ringing, they can be answered
and never drop. The thing is that there is no audio on any side! But I don’t
get any error or warning from JavaScript nor the Asterisk CLI. I’m using
Asterisk 12 + jsSIP.
If you could help me solving this I would be eternally greatful 😃 It’s for my
grade project …
These are my files:
sip.conf: http://pastebin.com/kWwXpi4V
http.conf: http://pastebin.com/ZwJWiiwf
SIP debugging for client REGISTER: http://pastebin.com/GNZETtQb
SIP debugging for extension call (Hello-World recording):
http://pastebin.com/0PxjLwBb
I followed these tutorials. If you have any other useful resource, I’d be glad
if you could share it:
http://stackoverflow.com/questions/26254980/websocket-connection-fails-with-asterisk-11
http://blog.gmc.uy/2014/04/asterisk-12-ubuntu-1204-pjproject-srtp.html
Furthermore, if I want to have a local Asterisk configuration, which should be
the IP address for the http.conf + DTLS certificates?? I tried with localhost
but RTP packets redirect to my eth IP.
Thanks in advance!!!!!!!!!!
De: Marek Červenka
Enviado el: martes, 15 de septiembre de 2015 06:37 a. m.
Para: [email protected]
hi,
i'm fighting with webrtc for 14 days
reporting my experience to minimize number of crazy asterisk users
i have working webrtc with simpl5 + asterisk 13 + pjproject 2.4.5 + chan_pjsip
+ secure websockets + secure audio + audio in both ways
problems
first, i needed run chan_sip for old hard phones and wss with chan_pjsip only
for webrtc. this is possible with patch from
https://issues.asterisk.org/jira/browse/ASTERISK-24106
chan_sip is not usable for webrtc because of
https://issues.asterisk.org/jira/browse/ASTERISK-24602
another problem arise with RTP/SAVPF negotiation
this can be solved with patch for Asterisk from
https://issues.asterisk.org/jira/browse/ASTERISK-24602
and for pjsip
http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2015-September/018607.html
i hope this info helps
what is your experience with WebRTC?
See you at WebRTC Expo Paris :)
p.s. many thanks to my colleague martin tomec for debugging support
p.s.2 relevant part of pjsip.conf
[global]
[transport-wss]
type=transport
protocol=wss ;udp,tcp,tls,ws,wss
bind=0.0.0.0
;===============ENDPOINT TEMPLATES
[endpoint-basic](!)
type=endpoint
transport=transport-wss
context=route_phones
disallow=all
allow=alaw
allow=ulaw
force_avp=yes
use_avpf=yes ; Determines whether res_pjsip will use and enforce usage of
media_encryption=dtls ; Determines whether res_pjsip will use and enforce
dtls_verify=no ; Verify that the provided peer certificate is valid (default:
dtls_rekey=0 ; Interval at which to renegotiate the TLS session and rekey
dtls_cert_file=/etc/pki/tls/certs/pbx.crt
dtls_private_key=/etc/pki/tls/private/pbx.key
dtls_setup=actpass
ice_support=yes ;This is specific to clients that support NAT traversal
media_use_received_transport=yes
[auth-userpass](!)
type=auth
auth_type=userpass
[aor-single-reg](!)
type=aor
remove_existing=yes
max_contacts=1
;===============DEVICES
[webrtc1](endpoint-basic)
auth=webrtc1
aors=webrtc1
[webrtc1](auth-userpass)
password=secret
username=webrtc1
[webrtc1](aor-single-reg)
relevant part of http.conf
[general]
enabled=yes
bindaddr=0.0.0.0
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/pki/tls/certs/pbx.crt
tlsprivatekey=/etc/pki/tls/private/pbx.key
--
---------------------------------------
Marek Cervenka
=======================================
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