8 Jul
2014
8 Jul
'14
10:41 p.m.
Thanks for reply
So I attached 2 log files:
1 from chrome to ff with no audio
2 from ff to chrome with ok audio
About ip address:
1.1.1.1 - kamailio server with rtpengine
2.2.2.2 - asterisk media server.
It runs as
./rtpengine --ip=1.1.1.1 --listen-udp=127.0.0.1:7723 --listen-ng=
127.0.0.1:7724 --pidfile=/var/run/rtpengine.pid
ringing between 2 peers thats behind NATs on both sides
mycompany.device-1884 at 3.3.3.3 (at chrome 35.0.1916.153 m)
mycompany.device-1091 at 4.4.4.4 (at ff 30 )
so when ring from chrome to ff at asterisk I i see rtp debug that displays
this:
Got RTP packet from 1.1.1.1:30716 (type 00, seq 023879, ts 143979401,
len 000160)
Sent RTP packet to 1.1.1.1:30722 (type 00, seq 048102, ts 143979400,
len 000160)
Got RTP packet from 1.1.1.1:30716 (type 00, seq 023880, ts 143979561,
len 000160)
Sent RTP packet to 1.1.1.1:30722 (type 00, seq 048103, ts 143979560,
len 000160)
Got RTP packet from 1.1.1.1:30716 (type 00, seq 023881, ts 143979721,
len 000160)
Sent RTP packet to 1.1.1.1:30722 (type 00, seq 048104, ts 143979720,
len 000160)
Got RTP packet from 1.1.1.1:30716 (type 00, seq 023882, ts 143979881,
len 000160)
Sent RTP packet to 1.1.1.1:30722 (type 00, seq 048105, ts 143979880,
len 000160)
so when ring from ff to chrome at asterisk I i see rtp debug that displays
this:
Got RTP packet from 1.1.1.1:30730 (type 00, seq 038011, ts 146424353,
len 000160)
Sent RTP packet to 1.1.1.1:30724 (type 00, seq 038427, ts 146424352,
len 000160)
Got RTP packet from 1.1.1.1:30730 (type 00, seq 038012, ts 146424513,
len 000160)
Sent RTP packet to 1.1.1.1:30724 (type 00, seq 038428, ts 146424512,
len 000160)
Got RTP packet from 1.1.1.1:30724 (type 00, seq 011839, ts 1249285912,
len 000160)
Got RTP packet from 1.1.1.1:30724 (type 00, seq 011840, ts 1249286072,
len 000160)
Got RTP packet from 1.1.1.1:30724 (type 00, seq 011841, ts 1249286232,
len 000160)
Sent RTP packet to 1.1.1.1:30730 (type 00, seq 037575, ts 1249285912,
len 000160)
Sent RTP packet to 1.1.1.1:30730 (type 00, seq 037576, ts 1249286072,
len 000160)
Sent RTP packet to 1.1.1.1:30730 (type 00, seq 037577, ts 1249286232,
len 000160)
Got RTP packet from 1.1.1.1:30730 (type 00, seq 038013, ts 146424673,
len 000160)
Sent RTP packet to 1.1.1.1:30724 (type 00, seq 038429, ts 146424672,
len 000160)
Got RTP packet from 1.1.1.1:30724 (type 00, seq 011842, ts 1249286392,
len 000160)
Sent RTP packet to 1.1.1.1:30730 (type 00, seq 037578, ts 1249286392,
len 000160)
We we use jssip.net demo from theys site.
Thanks very match and sorry for my bad English.
P. S. Log files also have kamailio log messages, if you do not need it just
do not pay attension to this.
2014-07-08 18:01 GMT+04:00 Richard Fuchs <rfuchs(a)sipwise.com>om>:
On 07/07/14 06:40 PM, Yuriy Gorlichenko wrote:
Hello. I have Kamailio
4.1.3+rtpengine_rtpproxy-ng as module for
rtpengine.
Kamailio installed as frontend (registration, auth, proxy ) of
asterisksk servers.
WEBRTC users registred at kamailio and asterisk works as media server.
When I try to call from Jssip from Firefox to chrome to way audio is fine.
When I call from chrome - I see rtp packets only from firefox. Not from
chrome.
At kamailio log when I call from chrome log the same as whe i try to
call from firefox (I can not see anithing wrong)
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.
cheers
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users