Thanks for this Mike. I'm going to test with verto (using sip.js currently.) other responses inline. On Thu, May 28, 2015 at 10:29 PM, Michael Jerris <mike@jerris.com> wrote:
On Thursday, May 28, 2015, Chris Mandra <mandra@gmail.com> wrote:
Hi Ken, thanks for the reply. Does https://conference.freeswitch.org/verto/ run fs_video? I kind of thought it was running an older version.
This is always running current code daily. We always dogfood our bleeding edge code.
good to know. thanks!
I'm hoping to get some help from the community bc, well, I'm probably not the only person with these problems and, like many people, I can't really afford paid assistance.
I have a few more questions:
I'm getting the followingf messages I could use some clarification on: 1. 2015-05-28 21:28:19.228386 [NOTICE] switch_core_media.c:3315 Asked by candidate to set remote rtcp audio addr to 205.215.241.76:59980 but this is rtcp-mux so no thanks (is this affecting anything?)
Your using chrome canary I think. This is a bug in canary. Our latest code addresses this chrome bug.
great.
2. 2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing video DTLS state from OFF to HANDSHAKE - it never gets beyond handshake - any ideas
Look at packet captures, maybe firewall blocking things?
I'll check.
3. 2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
packet 0 bytes (ice not ready @ line 4241!) - ice not ready - what's causing this?
This is because you don't have a dtls handshake yet.
makes sense
2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:404 Looking for
zrtp-hash 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:359 Deciding whether to pass zrtp-hash between legs 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:361 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
Should we be using ZRTP? if it weren't enabled at all would I see these messages?
Is one leg of your call passing in zrtp stuff ? You would not be running zrtp on a webrtc call as no webrtc end points I know of support it.
I don't think so, but I see messages abut it. I've seen docs mention it has to be turned on in the build - I haven't done that. Does it default to off still?
Lastly, it looks like we're not getting any sip responses from FS - Any
ideas why?
Maybe firewall, maybe misconfigured ip or acl? If you are using verto, where does sip come in to play?
using sip.js and not verto. will try the fs demo with verto.
All answers/help/hints are appreciated.
chris
On Thursday, May 28, 2015, Ken Rice <krice@freeswitch.org> wrote:
Hi Chris,
There is always https://conference.freeswitch.org/verto/ where we have been demoing this for quite a while
If you need this working like now I would suggest emailing consulting@freeswitch.org where you can get paid realtime assistance from one of the Core Dev Team
See you @ ClueCon! Ken
*http://www.ClueCon.com <http://www.ClueCon.com> http://www.freeswitch.org <http://www.freeswitch.org> http://www.OSTAG.org <http://www.OSTAG.org> *irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
On 5/28/15, 5:20 PM, "Chris Mandra" <mandra@gmail.com> wrote:
Hi guys. I had been able to use webrtc via chrome with FS 1.4 for connections and confenerce calls without incident (although the bandwidth and video quality seemed rather low as compared to the apprtc.appspot.com <http://apprtc.appspot.com> demo.) I have followed the instructions as well I am able on https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video but so far I can't make an actual connection; the certs pass the cert test without issue, but I have no joy using webrtc and fs_video
a few questions: 1. Is there any known problem using wildcard certs from godaddy? 2. is there other configuration beyond what's on the web page needed to make webrtc work? (I have the apropos ports open etc) 3. Are there any known issues with fs_video and sip.js? 4. I feel like it's getting hung op on dtls, as it never passes "2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS state from OFF to HANDSHAKE" (it repeats that twice with: "2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video packet 0 bytes (ice not ready @ line 4241!)" not long after - does that speak to something obvious to you guys? 5. is there a working webrtc / fs_video demo I can look at?
I realize this may all be solved by the release of 1.6, but I need this working now. I've been looking forward to this bc I know there have been some issues with webrtc, rtcp, freeswitch and solid BWE.
Thanks for all you do, chris
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