Hello,
On 23.12.17 21:05, Wilkins, Steve wrote: > > First I want to give Denys a huge shout-out for all of the help he has > given me. It is wonderful that boards like this exists and people are > so willing to help a newbie learn. > > > > I am on what I am hoping is my last major issue with WebRTCóWebRTC > calls (using tryit-jssip Chrome or Firefox). > > > > I am using Kamailio 5, and Asterisk 15 (pjsip). > > I am making calls between two WebRTC clients - Client1, and Client2 > (using tryit-jssip) > > > > Problem: If Client1 calls Client2, and Client2 ‘ANSWERS’, I only > have audio/video on Client1. Client2 gets no audio/video, but is > connected. If I switch things up and call Client1 from Client2, the > same thing happens (Client2 has audio/video and Client1 does not); I > can only get audio/video on the calling laptop; the called laptop has > no audio/video, but is connected. I see no errors in any of the logs. > > > > I am hoping that someone out there has seen this behavior before and > has an idea as to the cause and possible solution. > > > Are the clients behind the NAT? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
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