<div dir="ltr"><div>Thanks for reply</div><div>So I attached 2 log files:</div><div>1 from chrome to ff with no audio</div><div>2 from ff to chrome with ok audio</div><div><br></div><div>About ip address:</div><div>1.1.1.1 - kamailio server with rtpengine</div>
<div>2.2.2.2 - asterisk media server.</div><div>It runs as </div><div>./rtpengine --ip=1.1.1.1 --listen-udp=<a href="http://127.0.0.1:7723">127.0.0.1:7723</a> --listen-ng=<a href="http://127.0.0.1:7724">127.0.0.1:7724</a> --pidfile=/var/run/rtpengine.pid</div>
<div><br></div><div>ringing between 2 peers thats behind NATs on both sides</div><div>mycompany.device-1884 at 3.3.3.3 (at chrome 35.0.1916.153 m)</div><div>mycompany.device-1091 at 4.4.4.4 (at ff 30 )</div><div><br></div>
<div>so when ring from chrome to ff at asterisk I i see rtp debug that displays this:</div><div><br></div><div>Got RTP packet from <a href="http://1.1.1.1:30716">1.1.1.1:30716</a> (type 00, seq 023879, ts 143979401, len 000160)</div>
<div>Sent RTP packet to <a href="http://1.1.1.1:30722">1.1.1.1:30722</a> (type 00, seq 048102, ts 143979400, len 000160)</div><div>Got RTP packet from <a href="http://1.1.1.1:30716">1.1.1.1:30716</a> (type 00, seq 023880, ts 143979561, len 000160)</div>
<div>Sent RTP packet to <a href="http://1.1.1.1:30722">1.1.1.1:30722</a> (type 00, seq 048103, ts 143979560, len 000160)</div><div>Got RTP packet from <a href="http://1.1.1.1:30716">1.1.1.1:30716</a> (type 00, seq 023881, ts 143979721, len 000160)</div>
<div>Sent RTP packet to <a href="http://1.1.1.1:30722">1.1.1.1:30722</a> (type 00, seq 048104, ts 143979720, len 000160)</div><div>Got RTP packet from <a href="http://1.1.1.1:30716">1.1.1.1:30716</a> (type 00, seq 023882, ts 143979881, len 000160)</div>
<div>Sent RTP packet to <a href="http://1.1.1.1:30722">1.1.1.1:30722</a> (type 00, seq 048105, ts 143979880, len 000160)</div><div><br></div><div><br></div><div>so when ring from ff to chrome at asterisk I i see rtp debug that displays this:</div>
<div><br></div><div>Got RTP packet from <a href="http://1.1.1.1:30730">1.1.1.1:30730</a> (type 00, seq 038011, ts 146424353, len 000160)</div><div>Sent RTP packet to <a href="http://1.1.1.1:30724">1.1.1.1:30724</a> (type 00, seq 038427, ts 146424352, len 000160)</div>
<div>Got RTP packet from <a href="http://1.1.1.1:30730">1.1.1.1:30730</a> (type 00, seq 038012, ts 146424513, len 000160)</div><div>Sent RTP packet to <a href="http://1.1.1.1:30724">1.1.1.1:30724</a> (type 00, seq 038428, ts 146424512, len 000160)</div>
<div>Got RTP packet from <a href="http://1.1.1.1:30724">1.1.1.1:30724</a> (type 00, seq 011839, ts 1249285912, len 000160)</div><div>Got RTP packet from <a href="http://1.1.1.1:30724">1.1.1.1:30724</a> (type 00, seq 011840, ts 1249286072, len 000160)</div>
<div>Got RTP packet from <a href="http://1.1.1.1:30724">1.1.1.1:30724</a> (type 00, seq 011841, ts 1249286232, len 000160)</div><div>Sent RTP packet to <a href="http://1.1.1.1:30730">1.1.1.1:30730</a> (type 00, seq 037575, ts 1249285912, len 000160)</div>
<div>Sent RTP packet to <a href="http://1.1.1.1:30730">1.1.1.1:30730</a> (type 00, seq 037576, ts 1249286072, len 000160)</div><div>Sent RTP packet to <a href="http://1.1.1.1:30730">1.1.1.1:30730</a> (type 00, seq 037577, ts 1249286232, len 000160)</div>
<div>Got RTP packet from <a href="http://1.1.1.1:30730">1.1.1.1:30730</a> (type 00, seq 038013, ts 146424673, len 000160)</div><div>Sent RTP packet to <a href="http://1.1.1.1:30724">1.1.1.1:30724</a> (type 00, seq 038429, ts 146424672, len 000160)</div>
<div>Got RTP packet from <a href="http://1.1.1.1:30724">1.1.1.1:30724</a> (type 00, seq 011842, ts 1249286392, len 000160)</div><div>Sent RTP packet to <a href="http://1.1.1.1:30730">1.1.1.1:30730</a> (type 00, seq 037578, ts 1249286392, len 000160)</div>
<div><br></div><div><br></div><div>We we use <a href="http://jssip.net">jssip.net</a> demo from theys site.</div><div><br></div><div>Thanks very match and sorry for my bad English.</div><div><br></div><div>P. S. Log files also have kamailio log messages, if you do not need it just do not pay attension to this.</div>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">2014-07-08 18:01 GMT+04:00 Richard Fuchs <span dir="ltr"><<a href="mailto:rfuchs@sipwise.com" target="_blank">rfuchs@sipwise.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class="">On 07/07/14 06:40 PM, Yuriy Gorlichenko wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hello. I have Kamailio 4.1.3+rtpengine_rtpproxy-ng as module for rtpengine.<br>
<br>
Kamailio installed as frontend (registration, auth, proxy ) of<br>
asterisksk servers.<br>
<br>
WEBRTC users registred at kamailio and asterisk works as media server.<br>
<br>
When I try to call from Jssip from Firefox to chrome to way audio is fine.<br>
When I call from chrome - I see rtp packets only from firefox. Not from<br>
chrome.<br>
<br>
At kamailio log when I call from chrome log the same as whe i try to<br>
call from firefox (I can not see anithing wrong)<br>
</blockquote>
<br></div>
It would help to have the complete log messages from rtpengine. Make sure you have a very recent version of Firefox, it used to have certain WebRTC implementation problems.<br>
<br>
cheers<br>
<br>
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</blockquote></div><br></div>