Try looking at a pcap and comparing to the android logs. Are they sending the stun requests to the right place and are they getting to freeswitch? On Nov 27, 2013, at 7:38 PM, James Mortensen <james.mortensen@synclio.com> wrote:
The issue is that Android is showing STUN ping timeouts in the libjingle logs. This doesn't happen with Asterisk, so we just asked ourselves, "What is different in the SDP's" and we noticed it's the way RTCP is handled....
Now, I'm not suggesting using the port +1 method would solve the problem; it was mostly just something we thought to try to see if we'd get different results. If it's not worth trying, we'll try something else. :)
I don't have the Android debug logs handy at this time, but we'll work on putting them together.
Thanks! James
On Wed, Nov 27, 2013 at 4:14 PM, Michael Jerris <mike@jerris.com> wrote: We are using rtcp mux where it uses the same port. This is the default for webrtc and I don't think we have a way to disable it for outbound invites for webrtc media, but there shouldn't ever be a reason to disable it as all the browsers support it and its required by the spec. As for why its not working with chrome for android, I'm not sure, do you have a debug log of the call?
On Nov 27, 2013, at 5:56 PM, James Mortensen <james.mortensen@synclio.com> wrote:
Hello,
I've searched the list and documentation, and it's not yet clear if FreeSWITCH is able to send candidates in the same manner as we were doing with Asterisk 1.5 where it sends the RTCP to the RTP port +1. In other words, I'd see candidates that looked like this:
a=candidate:3441596188 1 udp 659136 10.3.1.1 20622 typ host generation 0 a=candidate:3441596188 1 udp 659136 50.18.X.X 20622 typ srflx generation 0 a=candidate:3441596188 2 udp 659136 10.3.1.1 20623 typ host generation 0 a=candidate:3441596188 2 udp 659136 50.18.X.X 20623 typ srflx generation 0
For instance, if the RTP port was 20622, the RTCP port would be 20623. We're getting no audio when trying to make Chrome to Android and Android to Android WebRTC calls, and we thought we'd try a different RTCP method as we did have it working with Asterisk but want to use FreeSWITCH instead.
I've checked out the latest FreeSWITCH code from the master branch.
Not sure if FreeSWITCH can do this yet, but if anyone knows that would be helpful.
Thank you! James