WEBRTC was working with 1.4, having no luck with fs_video
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 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
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.freeswitch.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH
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. 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?) 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 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? 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? Lastly, it looks like we're not getting any sip responses from FS - Any ideas why? 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
------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Ken
*http://www.FreeSWITCH.org <http://www.FreeSWITCH.org> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org <http://www.OSTAG.org> *irc.freenode.net #freeswitch Twitter: @FreeSWITCH
https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video On Fri, May 29, 2015 at 3:42 AM, 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.
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?)
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
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?
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?
Lastly, it looks like we're not getting any sip responses from FS - Any ideas why?
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
------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Ken
*http://www.FreeSWITCH.org <http://www.FreeSWITCH.org> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org <http://www.OSTAG.org> *irc.freenode.net #freeswitch Twitter: @FreeSWITCH
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video Please follow the instruction exactly, step by step, without any difference. (eg: use Debian Jessie, and copy and paste from the page). That will work.
From there, you can begin doing modifications.
On Fri, May 29, 2015 at 3:47 AM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
On Fri, May 29, 2015 at 3:42 AM, 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.
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?)
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
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?
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?
Lastly, it looks like we're not getting any sip responses from FS - Any ideas why?
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
------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Ken
*http://www.FreeSWITCH.org <http://www.FreeSWITCH.org> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org <http://www.OSTAG.org> *irc.freenode.net #freeswitch Twitter: @FreeSWITCH
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
-- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
Thanks Giovanni - I followed those instructions this afternoon. I feel like the main issues are 1. never getting beyond dtls handhsake: 2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS state from OFF to HANDSHAKE 2. not getting sip messages back from freeswitch ( we're using sip.js and not verto. This was all working (more or less) with 1.4) thanks, chris On Thu, May 28, 2015 at 9:47 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
On Fri, May 29, 2015 at 3:42 AM, 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.
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?)
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
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?
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?
Lastly, it looks like we're not getting any sip responses from FS - Any ideas why?
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
------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Ken
*http://www.FreeSWITCH.org <http://www.FreeSWITCH.org> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org <http://www.OSTAG.org> *irc.freenode.net #freeswitch Twitter: @FreeSWITCH
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- mandra c:410.258.5281
On Fri, May 29, 2015 at 4:02 AM, Chris Mandra <mandra@gmail.com> wrote:
Thanks Giovanni - I followed those instructions this afternoon. I feel like the main issues are 1. never getting beyond dtls handhsake:
2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS state from OFF to HANDSHAKE
2. not getting sip messages back from freeswitch ( we're using sip.js and not verto. This was all working (more or less) with 1.4)
it works with verto? make it first work with verto, so you can be sure all the infrastructure is there, then go for sip.js
thanks,
chris
On Thu, May 28, 2015 at 9:47 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
On Fri, May 29, 2015 at 3:42 AM, 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.
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?)
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
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?
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?
Lastly, it looks like we're not getting any sip responses from FS - Any ideas why?
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
------------------------------
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Ken
*http://www.FreeSWITCH.org <http://www.FreeSWITCH.org> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org <http://www.OSTAG.org> *irc.freenode.net #freeswitch Twitter: @FreeSWITCH
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- mandra c:410.258.5281
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
Conference.freeswitch.org has been running this code for a while and its used quite heavily. We use it every week for the past month or so for video with ClueCon Weekly. The point of paid assistance is you said you have to have this working now. The best way to get it working right now is to get paid support. Each of the following things could be any number of issues. 1. did you follow the setup documentation on Debian Jessie to the letter? 2. Are you using the Latest Chrome? (FireFox may work but no guarentees, IE and Safari don¹t even support WebRTC) 3. theres not enough information to even start saying anything about FS not responding to SIP messages... You¹ll have to review the logs and configs to figure out why 4. As far as ZRTP goes do you need it? We cant answer that, only you can... If you don¹t need it you can disable it On 5/28/15, 8:42 PM, "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.
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 <http://205.215.241.76:59980> but this is rtcp-mux so no thanks (is this affecting anything?)
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 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?
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?
Lastly, it looks like we're not getting any sip responses from FS - Any ideas why?
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 <http://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.freeswitch.org http://www.OSTAG.org irc.freenode.net <http://irc.freenode.net> #freeswitch Twitter: @FreeSWITCH @ClueCon
On 5/28/15, 5:20 PM, "Chris Mandra" <mandra@gmail.com <http://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> <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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <http://consulting@freeswitch.org> http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <http://FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH
Thanks for the thoughtful response Ken. Answers inline. On Thu, May 28, 2015 at 10:21 PM, Ken Rice <krice@freeswitch.org> wrote:
Conference.freeswitch.org has been running this code for a while and its used quite heavily. We use it every week for the past month or so for video with ClueCon Weekly.
The point of paid assistance is you said you have to have this working now. The best way to get it working right now is to get paid support.
Each of the following things could be any number of issues.
1. did you follow the setup documentation on Debian Jessie to the letter?
yup
1. 2. Are you using the Latest Chrome? (FireFox may work but no guarentees, IE and Safari don’t even support WebRTC)
yup
1. 2. theres not enough information to even start saying anything about FS not responding to SIP messages... You’ll have to review the logs and configs to figure out why
ok - I'm going to try using the FS verto demo to troubleshoot and see if
my problems are with sip.js
1. 2. As far as ZRTP goes do you need it? We cant answer that, only you can... If you don’t need it you can disable it
Prolly not. It has to be explicitly compiled into the build, right?
On 5/28/15, 8:42 PM, "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.
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 < http://205.215.241.76:59980> but this is rtcp-mux so no thanks (is this affecting anything?)
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 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?
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?
Lastly, it looks like we're not getting any sip responses from FS - Any ideas why?
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 <http://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 <http://irc.freenode.net> #freeswitch Twitter: @FreeSWITCH @ClueCon
On 5/28/15, 5:20 PM, "Chris Mandra" <mandra@gmail.com < http://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> <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
------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <http://consulting@freeswitch.org> http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org < http://FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Ken
*http://www.FreeSWITCH.org <http://www.FreeSWITCH.org> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org <http://www.OSTAG.org> *irc.freenode.net #freeswitch Twitter: @FreeSWITCH
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- mandra c:410.258.5281
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.
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.
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?
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.
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.
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?
All answers/help/hints are appreciated.
chris
On Thursday, May 28, 2015, Ken Rice <krice@freeswitch.org <javascript:_e(%7B%7D,'cvml','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
------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Ken
*http://www.FreeSWITCH.org <http://www.FreeSWITCH.org> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org <http://www.OSTAG.org> *irc.freenode.net #freeswitch Twitter: @FreeSWITCH
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
------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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-- Ken
*http://www.FreeSWITCH.org <http://www.FreeSWITCH.org> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org <http://www.OSTAG.org> *irc.freenode.net #freeswitch Twitter: @FreeSWITCH
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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-- mandra c:410.258.5281
yes still defaults to off, but I think we still do pass through even when it's built without it On Friday, May 29, 2015, Chris Mandra <mandra@gmail.com> wrote:
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 <javascript:_e(%7B%7D,'cvml','mike@jerris.com');>> wrote:
On Thursday, May 28, 2015, Chris Mandra <mandra@gmail.com <javascript:_e(%7B%7D,'cvml','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
------------------------------
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Ken
*http://www.FreeSWITCH.org <http://www.FreeSWITCH.org> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org <http://www.OSTAG.org> *irc.freenode.net #freeswitch Twitter: @FreeSWITCH
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');> http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users@lists.freeswitch.org');> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- mandra c:410.258.5281
participants (4)
-
Chris Mandra
-
Giovanni Maruzzelli
-
Ken Rice
-
Michael Jerris