VoIP Mailing List Archives

[cervajs at fpf.slu.cz]

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

[cervajs at fpf.slu.cz]

Dne 15.9.2015 v 13:37 Marek Červenka napsal(a):

[asandovalros at gmail.com]

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 (cervajs@fpf.slu.cz)
Enviado el: ‎martes‎, ‎15‎ de ‎septiembre‎ de ‎2015 ‎06‎:‎37‎ ‎a. m.
Para: asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)



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

[lists at jttech.se]

Den 2015-09-15 kl. 16:52, skrev asandovalros@gmail.com: