Hi, I've been testing a bit with WebRTC video using Chrome on Mac and a Freeswitch conference. The problem that I have is that the WebRTC client (Chrome) sends video keyframes every 3000 frames, which means that when a new participant joins the conference it can be some time before he/she gets a keyframe. The same problem occurs when the floor changes and the video stream source of the conference is switched to another participant, whose WebRTC client won't necessarily send a keyframe on floor change. As far as I understood after reading through a couple of Chrome tickets, a keyframe can be requested by the receiving party - in this case a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture Loss Indication) packet. Ticket FS-5596 mentions FIR in its description, so it seems as if FS already has support for FIR (or maybe I got it all wrong and the case described in this ticket refers to RTCP packets, which are simply being proxied by FS). Additionally FS indicates in the SDP that it supports FIR. BTW, I am not even sure if I have configured the RTCP in Freeswitch correctly. What I did was to uncommented "rtcp-video-interval-msec" and "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see in Chrome's log: webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR received. webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in RTCP RR extended highest sequence number. The real question is if the conference app can be configured/patched to send an RTCP FIR packet to the member who has the floor whenever a new participant joins the conference or in case the floor itself just moved to someone else. Has anyone tested something similar? BR Hristo
Please make sure you’re using the latest code, care to say what Rev you’re on? -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 28, 2014, at 10:13 AM, Hristo Trendev <dist.lists@gmail.com> wrote:
Hi,
I've been testing a bit with WebRTC video using Chrome on Mac and a Freeswitch conference. The problem that I have is that the WebRTC client (Chrome) sends video keyframes every 3000 frames, which means that when a new participant joins the conference it can be some time before he/she gets a keyframe. The same problem occurs when the floor changes and the video stream source of the conference is switched to another participant, whose WebRTC client won't necessarily send a keyframe on floor change.
As far as I understood after reading through a couple of Chrome tickets, a keyframe can be requested by the receiving party - in this case a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture Loss Indication) packet.
Ticket FS-5596 mentions FIR in its description, so it seems as if FS already has support for FIR (or maybe I got it all wrong and the case described in this ticket refers to RTCP packets, which are simply being proxied by FS). Additionally FS indicates in the SDP that it supports FIR.
BTW, I am not even sure if I have configured the RTCP in Freeswitch correctly. What I did was to uncommented "rtcp-video-interval-msec" and "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see in Chrome's log: webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR received. webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in RTCP RR extended highest sequence number.
The real question is if the conference app can be configured/patched to send an RTCP FIR packet to the member who has the floor whenever a new participant joins the conference or in case the floor itself just moved to someone else.
Has anyone tested something similar?
BR Hristo
Hi Brian, It's the latest v1.4.beta branch (555ef59) compiled on Debian 7.4. I can test with the master branch on Monday if needed. However, I am not really sure if the functionality to request a keyframe via RTCP FIR is even present in conference/freeswitch. In case it is indeed missing, then it will at least explain the behavior I see, otherwise I've probably missed something while testing. Best, Hristo On Fri, Mar 28, 2014 at 7:35 PM, Brian West <brian@freeswitch.org> wrote:
Please make sure you're using the latest code, care to say what Rev you're on? -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Mar 28, 2014, at 10:13 AM, Hristo Trendev <dist.lists@gmail.com> wrote:
Hi,
I've been testing a bit with WebRTC video using Chrome on Mac and a Freeswitch conference. The problem that I have is that the WebRTC client (Chrome) sends video keyframes every 3000 frames, which means that when a new participant joins the conference it can be some time before he/she gets a keyframe. The same problem occurs when the floor changes and the video stream source of the conference is switched to another participant, whose WebRTC client won't necessarily send a keyframe on floor change.
As far as I understood after reading through a couple of Chrome tickets, a keyframe can be requested by the receiving party - in this case a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture Loss Indication) packet.
Ticket FS-5596 mentions FIR in its description, so it seems as if FS already has support for FIR (or maybe I got it all wrong and the case described in this ticket refers to RTCP packets, which are simply being proxied by FS). Additionally FS indicates in the SDP that it supports FIR.
BTW, I am not even sure if I have configured the RTCP in Freeswitch correctly. What I did was to uncommented "rtcp-video-interval-msec" and "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see in Chrome's log: webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR received. webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in RTCP RR extended highest sequence number.
The real question is if the conference app can be configured/patched to send an RTCP FIR packet to the member who has the floor whenever a new participant joins the conference or in case the floor itself just moved to someone else.
Has anyone tested something similar?
BR Hristo
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
AFAIK FS has that so a jira would help 2014年3月29日 上午4:57于 "Hristo Trendev" <dist.lists@gmail.com>写道：
Hi Brian,
It's the latest v1.4.beta branch (555ef59) compiled on Debian 7.4. I can test with the master branch on Monday if needed. However, I am not really sure if the functionality to request a keyframe via RTCP FIR is even present in conference/freeswitch. In case it is indeed missing, then it will at least explain the behavior I see, otherwise I've probably missed something while testing.
Best, Hristo
On Fri, Mar 28, 2014 at 7:35 PM, Brian West <brian@freeswitch.org> wrote:
Please make sure you’re using the latest code, care to say what Rev you’re on? -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Mar 28, 2014, at 10:13 AM, Hristo Trendev <dist.lists@gmail.com> wrote:
Hi,
I've been testing a bit with WebRTC video using Chrome on Mac and a Freeswitch conference. The problem that I have is that the WebRTC client (Chrome) sends video keyframes every 3000 frames, which means that when a new participant joins the conference it can be some time before he/she gets a keyframe. The same problem occurs when the floor changes and the video stream source of the conference is switched to another participant, whose WebRTC client won't necessarily send a keyframe on floor change.
As far as I understood after reading through a couple of Chrome tickets, a keyframe can be requested by the receiving party - in this case a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture Loss Indication) packet.
Ticket FS-5596 mentions FIR in its description, so it seems as if FS already has support for FIR (or maybe I got it all wrong and the case described in this ticket refers to RTCP packets, which are simply being proxied by FS). Additionally FS indicates in the SDP that it supports FIR.
BTW, I am not even sure if I have configured the RTCP in Freeswitch correctly. What I did was to uncommented "rtcp-video-interval-msec" and "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see in Chrome's log: webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR received. webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in RTCP RR extended highest sequence number.
The real question is if the conference app can be configured/patched to send an RTCP FIR packet to the member who has the floor whenever a new participant joins the conference or in case the floor itself just moved to someone else.
Has anyone tested something similar?
BR Hristo
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
Fir is sent explicitly no settings req On Mar 28, 2014 5:15 PM, "Seven Du" <dujinfang@gmail.com> wrote:
AFAIK FS has that so a jira would help 2014年3月29日 上午4:57于 "Hristo Trendev" <dist.lists@gmail.com>写道：
Hi Brian,
It's the latest v1.4.beta branch (555ef59) compiled on Debian 7.4. I can test with the master branch on Monday if needed. However, I am not really sure if the functionality to request a keyframe via RTCP FIR is even present in conference/freeswitch. In case it is indeed missing, then it will at least explain the behavior I see, otherwise I've probably missed something while testing.
Best, Hristo
On Fri, Mar 28, 2014 at 7:35 PM, Brian West <brian@freeswitch.org> wrote:
Please make sure you’re using the latest code, care to say what Rev you’re on? -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Mar 28, 2014, at 10:13 AM, Hristo Trendev <dist.lists@gmail.com> wrote:
Hi,
I've been testing a bit with WebRTC video using Chrome on Mac and a Freeswitch conference. The problem that I have is that the WebRTC client (Chrome) sends video keyframes every 3000 frames, which means that when a new participant joins the conference it can be some time before he/she gets a keyframe. The same problem occurs when the floor changes and the video stream source of the conference is switched to another participant, whose WebRTC client won't necessarily send a keyframe on floor change.
As far as I understood after reading through a couple of Chrome tickets, a keyframe can be requested by the receiving party - in this case a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture Loss Indication) packet.
Ticket FS-5596 mentions FIR in its description, so it seems as if FS already has support for FIR (or maybe I got it all wrong and the case described in this ticket refers to RTCP packets, which are simply being proxied by FS). Additionally FS indicates in the SDP that it supports FIR.
BTW, I am not even sure if I have configured the RTCP in Freeswitch correctly. What I did was to uncommented "rtcp-video-interval-msec" and "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see in Chrome's log: webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR received. webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in RTCP RR extended highest sequence number.
The real question is if the conference app can be configured/patched to send an RTCP FIR packet to the member who has the floor whenever a new participant joins the conference or in case the floor itself just moved to someone else.
Has anyone tested something similar?
BR Hristo
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
Hi, One of the browsers that I used for the initial testing was Chrome on Android 4.3. I guess this was causing some of the problems. I retested with the latest version from master (0876df6). When I make the test with two desktop browsers indeed a new keyframe seems to be requested on every floor change, as in this case the video stream switches almost instantly. However, the problem is still present for anyone who has just joined a conference. Unless the floor changes shortly after someone joins, he has to wait for the next keyframe. This is really simple to reproduce if there is only one speaker in the conference and all the others who are joining the conference are muted by default - in this case there is virtually no chance of a floor change to occur. Is a FIR packet also sent when someone joins a conference or is it currently only sent on floor change events? Best Hristo On Sat, Mar 29, 2014 at 3:11 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Fir is sent explicitly no settings req On Mar 28, 2014 5:15 PM, "Seven Du" <dujinfang@gmail.com> wrote:
AFAIK FS has that so a jira would help 2014年3月29日 上午4:57于 "Hristo Trendev" <dist.lists@gmail.com>写道：
Hi Brian,
It's the latest v1.4.beta branch (555ef59) compiled on Debian 7.4. I can test with the master branch on Monday if needed. However, I am not really sure if the functionality to request a keyframe via RTCP FIR is even present in conference/freeswitch. In case it is indeed missing, then it will at least explain the behavior I see, otherwise I've probably missed something while testing.
Best, Hristo
On Fri, Mar 28, 2014 at 7:35 PM, Brian West <brian@freeswitch.org>wrote:
Please make sure you’re using the latest code, care to say what Rev you’re on? -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Mar 28, 2014, at 10:13 AM, Hristo Trendev <dist.lists@gmail.com> wrote:
Hi,
I've been testing a bit with WebRTC video using Chrome on Mac and a Freeswitch conference. The problem that I have is that the WebRTC client (Chrome) sends video keyframes every 3000 frames, which means that when a new participant joins the conference it can be some time before he/she gets a keyframe. The same problem occurs when the floor changes and the video stream source of the conference is switched to another participant, whose WebRTC client won't necessarily send a keyframe on floor change.
As far as I understood after reading through a couple of Chrome tickets, a keyframe can be requested by the receiving party - in this case a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture Loss Indication) packet.
Ticket FS-5596 mentions FIR in its description, so it seems as if FS already has support for FIR (or maybe I got it all wrong and the case described in this ticket refers to RTCP packets, which are simply being proxied by FS). Additionally FS indicates in the SDP that it supports FIR.
BTW, I am not even sure if I have configured the RTCP in Freeswitch correctly. What I did was to uncommented "rtcp-video-interval-msec" and "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see in Chrome's log: webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR received. webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in RTCP RR extended highest sequence number.
The real question is if the conference app can be configured/patched to send an RTCP FIR packet to the member who has the floor whenever a new participant joins the conference or in case the floor itself just moved to someone else.
Has anyone tested something similar?
BR Hristo
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
It did not but I just added a patch so it will. Try master. On Mon, Mar 31, 2014 at 10:52 AM, Hristo Trendev <dist.lists@gmail.com>wrote:
Hi,
One of the browsers that I used for the initial testing was Chrome on Android 4.3. I guess this was causing some of the problems.
I retested with the latest version from master (0876df6). When I make the test with two desktop browsers indeed a new keyframe seems to be requested on every floor change, as in this case the video stream switches almost instantly.
However, the problem is still present for anyone who has just joined a conference. Unless the floor changes shortly after someone joins, he has to wait for the next keyframe. This is really simple to reproduce if there is only one speaker in the conference and all the others who are joining the conference are muted by default - in this case there is virtually no chance of a floor change to occur.
Is a FIR packet also sent when someone joins a conference or is it currently only sent on floor change events?
Best Hristo
On Sat, Mar 29, 2014 at 3:11 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Fir is sent explicitly no settings req On Mar 28, 2014 5:15 PM, "Seven Du" <dujinfang@gmail.com> wrote:
AFAIK FS has that so a jira would help 2014年3月29日 上午4:57于 "Hristo Trendev" <dist.lists@gmail.com>写道：
Hi Brian,
It's the latest v1.4.beta branch (555ef59) compiled on Debian 7.4. I can test with the master branch on Monday if needed. However, I am not really sure if the functionality to request a keyframe via RTCP FIR is even present in conference/freeswitch. In case it is indeed missing, then it will at least explain the behavior I see, otherwise I've probably missed something while testing.
Best, Hristo
On Fri, Mar 28, 2014 at 7:35 PM, Brian West <brian@freeswitch.org>wrote:
Please make sure you’re using the latest code, care to say what Rev you’re on? -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Mar 28, 2014, at 10:13 AM, Hristo Trendev <dist.lists@gmail.com> wrote:
Hi,
I've been testing a bit with WebRTC video using Chrome on Mac and a Freeswitch conference. The problem that I have is that the WebRTC client (Chrome) sends video keyframes every 3000 frames, which means that when a new participant joins the conference it can be some time before he/she gets a keyframe. The same problem occurs when the floor changes and the video stream source of the conference is switched to another participant, whose WebRTC client won't necessarily send a keyframe on floor change.
As far as I understood after reading through a couple of Chrome tickets, a keyframe can be requested by the receiving party - in this case a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture Loss Indication) packet.
Ticket FS-5596 mentions FIR in its description, so it seems as if FS already has support for FIR (or maybe I got it all wrong and the case described in this ticket refers to RTCP packets, which are simply being proxied by FS). Additionally FS indicates in the SDP that it supports FIR.
BTW, I am not even sure if I have configured the RTCP in Freeswitch correctly. What I did was to uncommented "rtcp-video-interval-msec" and "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see in Chrome's log: webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR received. webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in RTCP RR extended highest sequence number.
The real question is if the conference app can be configured/patched to send an RTCP FIR packet to the member who has the floor whenever a new participant joins the conference or in case the floor itself just moved to someone else.
Has anyone tested something similar?
BR Hristo
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Thanks Anthony! It works now. On Mon, Mar 31, 2014 at 7:39 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
It did not but I just added a patch so it will. Try master.
On Mon, Mar 31, 2014 at 10:52 AM, Hristo Trendev <dist.lists@gmail.com>wrote:
Hi,
One of the browsers that I used for the initial testing was Chrome on Android 4.3. I guess this was causing some of the problems.
I retested with the latest version from master (0876df6). When I make the test with two desktop browsers indeed a new keyframe seems to be requested on every floor change, as in this case the video stream switches almost instantly.
However, the problem is still present for anyone who has just joined a conference. Unless the floor changes shortly after someone joins, he has to wait for the next keyframe. This is really simple to reproduce if there is only one speaker in the conference and all the others who are joining the conference are muted by default - in this case there is virtually no chance of a floor change to occur.
Is a FIR packet also sent when someone joins a conference or is it currently only sent on floor change events?
Best Hristo
On Sat, Mar 29, 2014 at 3:11 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Fir is sent explicitly no settings req On Mar 28, 2014 5:15 PM, "Seven Du" <dujinfang@gmail.com> wrote:
AFAIK FS has that so a jira would help 2014年3月29日 上午4:57于 "Hristo Trendev" <dist.lists@gmail.com>写道：
Hi Brian,
It's the latest v1.4.beta branch (555ef59) compiled on Debian 7.4. I can test with the master branch on Monday if needed. However, I am not really sure if the functionality to request a keyframe via RTCP FIR is even present in conference/freeswitch. In case it is indeed missing, then it will at least explain the behavior I see, otherwise I've probably missed something while testing.
Best, Hristo
On Fri, Mar 28, 2014 at 7:35 PM, Brian West <brian@freeswitch.org>wrote:
Please make sure you’re using the latest code, care to say what Rev you’re on? -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Mar 28, 2014, at 10:13 AM, Hristo Trendev <dist.lists@gmail.com> wrote:
> Hi, > > I've been testing a bit with WebRTC video using Chrome on Mac and a Freeswitch conference. The problem that I have is that the WebRTC client (Chrome) sends video keyframes every 3000 frames, which means that when a new participant joins the conference it can be some time before he/she gets a keyframe. The same problem occurs when the floor changes and the video stream source of the conference is switched to another participant, whose WebRTC client won't necessarily send a keyframe on floor change. > > As far as I understood after reading through a couple of Chrome tickets, a keyframe can be requested by the receiving party - in this case a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture Loss Indication) packet. > > Ticket FS-5596 mentions FIR in its description, so it seems as if FS already has support for FIR (or maybe I got it all wrong and the case described in this ticket refers to RTCP packets, which are simply being proxied by FS). Additionally FS indicates in the SDP that it supports FIR. > > BTW, I am not even sure if I have configured the RTCP in Freeswitch correctly. What I did was to uncommented "rtcp-video-interval-msec" and "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see in Chrome's log: > webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR received. > webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in RTCP RR extended highest sequence number. > > The real question is if the conference app can be configured/patched to send an RTCP FIR packet to the member who has the floor whenever a new participant joins the conference or in case the floor itself just moved to someone else. > > Has anyone tested something similar? > > BR > Hristo
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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participants (4)
-
Anthony Minessale
-
Brian West
-
Hristo Trendev
-
Seven Du