<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">This probably needs some work.. Can you open up a jira and get as much details as possible about how this should be negotiated, In the case you are talking about the timing of when we know what the bandwidth and resolution of the video could be probelmatic, in fact, they can change mid call too, so any info you have on capabilites of your device to support things such as rtcp tmmbr and such to adjust these settings is important. It may be as simple as what is passed in the sdp is the top bound, but this may depend on the device. Get as much detail into the jira as possible and we can discuss more there.<div class="">Thanks</div><div class="">Mike</div><div class=""><br class=""><div><blockquote type="cite" class=""><div class="">On Mar 18, 2016, at 2:02 PM, Volodymyr Fedorov <<a href="mailto:lexxua@gmail.com" class="">lexxua@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">Thanks Mike for answer, quite strange that freeswitch always answer with b=AS:1024 for video stream.<div class="">Let me describe Invite from UA (Grandstream gvc3200) SDP:</div><div class=""><br class=""></div><div class=""><div class="">v=0</div><div class="">o=ipod7 8000 8000 IN IP4 77.120.9.98</div><div class="">s=SIP Call</div><div class="">c=IN IP4 77.120.9.98</div><div class="">b=AS:1312</div><div class="">t=0 0</div><div class="">m=audio 5004 RTP/AVP 0 8 9 104 123 101</div><div class="">b=TIAS:64000</div><div class="">a=sendrecv</div><div class="">a=rtpmap:0 PCMU/8000</div><div class="">a=ptime:20</div><div class="">a=rtpmap:8 PCMA/8000</div><div class="">a=rtpmap:9 G722/8000</div><div class="">a=rtpmap:104 G7221/16000</div><div class="">a=fmtp:104 bitrate=32000</div><div class="">a=rtpmap:123 opus/48000/2</div><div class="">a=fmtp:123 useinbandfec=1; sprop-maxcapturerate=16000; stereo=0; sprop-stereo=0</div><div class="">a=rtpmap:101 telephone-event/8000</div><div class="">a=fmtp:101 0-15</div><div class="">m=video 5006 RTP/AVP 99</div><div class="">b=TIAS:1024000</div><div class="">a=sendrecv</div><div class="">a=rtpmap:99 H264/90000</div><div class="">a=fmtp:99 profile-level-id=42801F; packetization-mode=1</div><div class="">a=framerate:15</div><div class="">a=content:main</div><div class="">a=label:11</div></div><div class=""><br class=""></div><div class="">FS respond part:</div><div class=""><div class="">v=0</div><div class="">o=FreeSWITCH 1458296997 1458296998 IN IP4 77.121.1.26</div><div class="">s=FreeSWITCH</div><div class="">c=IN IP4 77.121.1.26</div><div class="">t=0 0</div><div class="">m=audio 26504 RTP/AVP 0 101</div><div class="">a=rtpmap:0 PCMU/8000</div><div class="">a=rtpmap:101 telephone-event/8000</div><div class="">a=fmtp:101 0-16</div><div class="">a=ptime:20</div><div class="">m=video 31298 RTP/AVP 99</div><div class="">b=AS:1024</div><div class="">a=rtpmap:99 H264/90000</div></div><div class=""><br class=""></div><div class="">What really confusing me are b=AS:1024 for Video rtp stream and absent a=fmtp:99 profile-level-id=42801F; packetization-mode=1 in freeswitch answer SDP. Why I`m interesting because my device always starts with video resolution<span style="color:rgb(38,50,56);font-size:13px;line-height:16px" class=""> 176x144. I tried with asterisk and it`s respond part looks like this for video:</span></div><div class=""><span style="color:rgb(38,50,56);font-size:13px;line-height:16px" class="">a=fmtp:99 profile-level-id=428028 and device starts to send 1920x1080. Now I`m using Freeswitch </span><font color="#263238" class=""><span style="line-height:16px" class="">Version 1.6.6 -13-d2d0b32 64bit from stable repo.</span></font><span style="color:rgb(38,50,56);font-size:13px;line-height:16px" class=""><br class=""></span></div><div class=""><br class=""></div></div><div class="gmail_extra"><br class=""><div class="gmail_quote">On Wed, Mar 16, 2016 at 9:17 PM, Michael Jerris <span dir="ltr" class=""><<a href="mailto:mike@jerris.com" target="_blank" class="">mike@jerris.com</a>></span> wrote:<br class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word" class="">Its only used in mod_conference, and I believe it will be adjusted via rtcp.<div class=""><br class=""><div class=""><blockquote type="cite" class=""><div class=""><div class="h5"><div class="">On Mar 14, 2016, at 4:21 AM, Volodymyr Fedorov <<a href="mailto:lexxua@gmail.com" target="_blank" class="">lexxua@gmail.com</a>> wrote:</div><br class=""></div></div><div class=""><div class=""><div class="h5"><div dir="ltr" class="">Hi, community!<br class="">In wiki page <a href="https://freeswitch.org/confluence/display/FREESWITCH/mod_conference" target="_blank" class="">https://freeswitch.org/confluence/display/FREESWITCH/mod_conference</a><br class="">I found parameter for conf room: video-codec-bandwidth . It will be used as SDP respond part as a Bandwidth ("b=") from Freeswitch ?<br class=""><div class=""><div class=""><div class=""><div class="">Thank you in advance</div>-- <br class=""><div class=""><div dir="ltr" class=""><div class=""><div style="font-size:small" class="">Best regards,</div><div style="font-size:small" class="">Volodymyr</div></div></div></div>
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<a href="http://www.freeswitch.org/" rel="noreferrer" target="_blank" class="">http://www.freeswitch.org</a><br class=""></blockquote></div><br class=""><br clear="all" class=""><div class=""><br class=""></div>-- <br class=""><div class="gmail_signature"><div dir="ltr" class=""><div class=""><div style="font-size:small" class="">Best regards,</div><div style="font-size:small" class="">Volodymyr</div></div></div></div>
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