search

doghell / webrtc2sip

Automatically exported from code.google.com/p/webrtc2sip
0 stars 0 forks source link

rtp packets not forwarded #32

Open GoogleCodeExporter opened 8 years ago

GoogleCodeExporter commented 8 years ago 
What steps will reproduce the problem?

I've successfully installed and run webrtc2sip on my debian6-64 and run it.
When I make a call session established and in tcdump I see packets comming to 
webrtc2sip from my chrome and from legacy sip soft, but no packets go out from 
webrtc2sip.
I enabled "Disable Video:" and "Enable RTCWeb Breaker" on expert page. Also in 
browser js script console there is no any errors.

What version of the product are you using? On what operating system?
webrtc2sip on debian6 -64, 
chrome 23.0.1271.95 m win7-64

Please provide any additional information below.
This ticket from topic 
https://groups.google.com/forum/?fromgroups=#!topic/doubango/DPQbDSGsUQ0

Original issue reported on code.google.com by teih...@gmail.com on 11 Dec 2012 at 11:39

Attachments:

GoogleCodeExporter commented 8 years ago 
The problem was in ptime as Gökhan Barış Aker said. Legacy SIP sends rtp 
data every 30 ms and when I've change it to 20 ms, everything become ok. So 
issue is about adapting ptime in webrtc2sip. Thank you.

Original comment by teih...@gmail.com on 25 Apr 2013 at 4:47