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<div data-externalstyle="false" dir="ltr" style="font-family: 'Calibri', 'Segoe UI', 'Meiryo', 'Microsoft YaHei UI', 'Microsoft JhengHei UI', 'Malgun Gothic', 'sans-serif';font-size:12pt;"><div>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 …</div><div><br></div><div>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.</div><div><br></div><div>If you could help me solving this I would be eternally greatful 😃 It’s for my grade project …<br>These are my files:</div><div>sip.conf: <a href="http://pastebin.com/kWwXpi4V" target="_parent">http://pastebin.com/kWwXpi4V</a></div><div>http.conf: http://pastebin.com/ZwJWiiwf</div><div>SIP debugging for client REGISTER: http://pastebin.com/GNZETtQb</div><div>SIP debugging for extension call (Hello-World recording): <a href="http://pastebin.com/0PxjLwBb" target="_parent">http://pastebin.com/0PxjLwBb</a><br><br>I followed these tutorials. If you have any other useful resource, I’d be glad if you could share it:<br><a href="http://stackoverflow.com/questions/26254980/websocket-connection-fails-with-asterisk-11" target="_parent">http://stackoverflow.com/questions/26254980/websocket-connection-fails-with-asterisk-11</a></div><div><a href="http://blog.gmc.uy/2014/04/asterisk-12-ubuntu-1204-pjproject-srtp.html" target="_parent">http://blog.gmc.uy/2014/04/asterisk-12-ubuntu-1204-pjproject-srtp.html</a></div><div><br>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. <br><br>Thanks in advance!!!!!!!!!! <br></div><div data-signatureblock="true"><br></div><div style="padding-top: 5px; border-top-color: rgb(229, 229, 229); border-top-width: 1px; border-top-style: solid;"><div><font face=" 'Calibri', 'Segoe UI', 'Meiryo', 'Microsoft YaHei UI', 'Microsoft JhengHei UI', 'Malgun Gothic', 'sans-serif'" style='line-height: 15pt; letter-spacing: 0.02em; font-family: "Calibri", "Segoe UI", "Meiryo", "Microsoft YaHei UI", "Microsoft JhengHei UI", "Malgun Gothic", "sans-serif"; font-size: 12pt;'><b>De:</b> <a href="mailto:cervajs@fpf.slu.cz" target="_parent">Marek Červenka</a><br><b>Enviado el:</b> martes, 15 de septiembre de 2015 06:37 a. m.<br><b>Para:</b> <a href="mailto:asterisk-users@lists.digium.com" target="_parent">asterisk-users@lists.digium.com</a></font></div></div><div><br></div><div dir="">
hi,<br>
<span>
<div><span><br>
</span></div>
<div><span>i'm fighting with webrtc for 14 days</span></div>
<div><span>reporting my experience to minimize number of crazy
asterisk users <br>
</span></div>
<div><span><br>
i
have working webrtc with simpl5 + asterisk 13 + pjproject
2.4.5 + chan_pjsip + secure websockets + secure audio + audio
in both ways<br>
<br>
problems<br>
first, i needed run chan_sip for old hard phones and wss with
chan_pjsip only for webrtc. this is possible with patch from<br>
<span style="-ms-word-wrap: break-word;"><a class="moz-txt-link-freetext" href="https://issues.asterisk.org/jira/browse/ASTERISK-24106" target="_parent">https://issues.asterisk.org/jira/browse/ASTERISK-24106</a></span><br>
<br>
chan_sip is not usable for webrtc because of<br>
</span></div>
</span>
<span>
<div><span><a class="moz-txt-link-freetext" href="https://issues.asterisk.org/jira/browse/ASTERISK-24602" target="_parent">https://issues.asterisk.org/jira/browse/ASTERISK-24602</a><br>
<br>
another problem arise with RTP/SAVPF negotiation<br>
this can be solved with patch for Asterisk from<br>
</span><span><span><span><a class="moz-txt-link-freetext" href="https://issues.asterisk.org/jira/browse/ASTERISK-24602" target="_parent">https://issues.asterisk.org/jira/browse/ASTERISK-24602</a></span></span><br>
and for pjsip<br>
<a class="moz-txt-link-freetext" href="http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2015-September/018607.html" target="_parent">http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2015-September/018607.html</a><br>
<br>
i hope this info helps<br>
<br>
what is your experience with WebRTC?<br>
<br>
See you at WebRTC Expo Paris :)<br>
<br>
p.s. many thanks to my colleague martin tomec for debugging
support<br>
</span></div>
</span>
<br>
p.s.2 relevant part of pjsip.conf<br>
<br>
[global]<br>
[transport-wss]<br>
type=transport<br>
protocol=wss ;udp,tcp,tls,ws,wss<br>
bind=0.0.0.0<br>
<br>
;===============ENDPOINT TEMPLATES<br>
<br>
[endpoint-basic](!)<br>
type=endpoint<br>
transport=transport-wss<br>
context=route_phones<br>
disallow=all<br>
allow=alaw<br>
allow=ulaw<br>
force_avp=yes<br>
use_avpf=yes ; Determines whether res_pjsip will use and enforce
usage of<br>
media_encryption=dtls ; Determines whether res_pjsip will use and
enforce<br>
dtls_verify=no ; Verify that the provided peer certificate is valid
(default:<br>
dtls_rekey=0 ; Interval at which to renegotiate the TLS session
and rekey<br>
dtls_cert_file=/etc/pki/tls/certs/pbx.crt<br>
dtls_private_key=/etc/pki/tls/private/pbx.key<br>
dtls_setup=actpass<br>
ice_support=yes ;This is specific to clients that support NAT
traversal<br>
media_use_received_transport=yes<br>
<br>
[auth-userpass](!)<br>
type=auth<br>
auth_type=userpass<br>
<br>
[aor-single-reg](!)<br>
type=aor<br>
remove_existing=yes<br>
max_contacts=1<br>
<br>
<br>
;===============DEVICES<br>
<br>
[webrtc1](endpoint-basic)<br>
auth=webrtc1<br>
aors=webrtc1<br>
<br>
[webrtc1](auth-userpass)<br>
password=secret<br>
username=webrtc1<br>
<br>
[webrtc1](aor-single-reg)<br>
<br>
relevant part of http.conf<br>
[general]<br>
enabled=yes<br>
bindaddr=0.0.0.0<br>
tlsenable=yes<br>
tlsbindaddr=0.0.0.0:8089<br>
tlscertfile=/etc/pki/tls/certs/pbx.crt<br>
tlsprivatekey=/etc/pki/tls/private/pbx.key<br>
<br>
<pre class="moz-signature">--
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Marek Cervenka
=======================================
</pre>
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