<div dir="ltr">The param in the SDP is the max bw the client should send or recv. Its controlled from channel variables or globals in vars.xml.<div>You should coordinate the value with what you want in your conference because the conference does not control it.</div><div><br></div><div>
<p class=""><span class="">rtp_video_max_bandwidth_in</span></p><p class=""><span class="">
</span></p><p class=""><span class="">rtp_video_max_bandwidth_out</span></p><p class=""><span class=""><br></span></p><p class=""><span class=""><br></span></p><p class=""><span class=""><br></span></p><div><br></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Mar 18, 2016 at 1:15 PM, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">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>Thanks</div><div>Mike</div><div><div class="h5"><div><br><div><blockquote type="cite"><div>On Mar 18, 2016, at 2:02 PM, Volodymyr Fedorov <<a href="mailto:lexxua@gmail.com" target="_blank">lexxua@gmail.com</a>> wrote:</div><br><div><div dir="ltr">Thanks Mike for answer, quite strange that freeswitch always answer with b=AS:1024 for video stream.<div>Let me describe Invite from UA (Grandstream gvc3200) SDP:</div><div><br></div><div><div>v=0</div><div>o=ipod7 8000 8000 IN IP4 77.120.9.98</div><div>s=SIP Call</div><div>c=IN IP4 77.120.9.98</div><div>b=AS:1312</div><div>t=0 0</div><div>m=audio 5004 RTP/AVP 0 8 9 104 123 101</div><div>b=TIAS:64000</div><div>a=sendrecv</div><div>a=rtpmap:0 PCMU/8000</div><div>a=ptime:20</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:9 G722/8000</div><div>a=rtpmap:104 G7221/16000</div><div>a=fmtp:104 bitrate=32000</div><div>a=rtpmap:123 opus/48000/2</div><div>a=fmtp:123 useinbandfec=1; sprop-maxcapturerate=16000; stereo=0; sprop-stereo=0</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-15</div><div>m=video 5006 RTP/AVP 99</div><div>b=TIAS:1024000</div><div>a=sendrecv</div><div>a=rtpmap:99 H264/90000</div><div>a=fmtp:99 profile-level-id=42801F; packetization-mode=1</div><div>a=framerate:15</div><div>a=content:main</div><div>a=label:11</div></div><div><br></div><div>FS respond part:</div><div><div>v=0</div><div>o=FreeSWITCH 1458296997 1458296998 IN IP4 77.121.1.26</div><div>s=FreeSWITCH</div><div>c=IN IP4 77.121.1.26</div><div>t=0 0</div><div>m=audio 26504 RTP/AVP 0 101</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=ptime:20</div><div>m=video 31298 RTP/AVP 99</div><div>b=AS:1024</div><div>a=rtpmap:99 H264/90000</div></div><div><br></div><div>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"> 176x144. I tried with asterisk and it`s respond part looks like this for video:</span></div><div><span style="color:rgb(38,50,56);font-size:13px;line-height:16px">a=fmtp:99 profile-level-id=428028 and device starts to send 1920x1080. Now I`m using Freeswitch </span><font color="#263238"><span style="line-height:16px">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"><br></span></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Mar 16, 2016 at 9:17 PM, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">Its only used in mod_conference, and I believe it will be adjusted via rtcp.<div><br><div><blockquote type="cite"><div><div><div>On Mar 14, 2016, at 4:21 AM, Volodymyr Fedorov <<a href="mailto:lexxua@gmail.com" target="_blank">lexxua@gmail.com</a>> wrote:</div><br></div></div><div><div><div><div dir="ltr">Hi, community!<br>In wiki page <a href="https://freeswitch.org/confluence/display/FREESWITCH/mod_conference" target="_blank">https://freeswitch.org/confluence/display/FREESWITCH/mod_conference</a><br>I found parameter for conf room: video-codec-bandwidth . It will be used as SDP respond part as a Bandwidth ("b=") from Freeswitch ?<br><div><div><div><div>Thank you in advance</div>-- <br><div><div dir="ltr"><div><div style="font-size:small">Best regards,</div><div style="font-size:small">Volodymyr</div></div></div></div>
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<a href="http://www.freeswitch.org/" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div><div dir="ltr"><div><div style="font-size:small">Best regards,</div><div style="font-size:small">Volodymyr</div></div></div></div>
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<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div class="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr">Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬<div><br><div>☞ <a href="http://freeswitch.org/" target="_blank">http://freeswitch.org/</a> ☞ <a href="http://cluecon.com/" target="_blank">http://cluecon.com/</a> ☞ <a href="http://twitter.com/FreeSWITCH" target="_blank">http://twitter.com/FreeSWITCH</a></div><div><div>☞ <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch ☞ <u><a href="http://freeswitch.org/g+" target="_blank">http://freeswitch.org/g+</a></u><br><br></div><div>ClueCon Weekly Development Call <br></div><div>☎ <a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a> ☎ +19193869900 </div><div><br></div></div></div><div><a href="https://www.youtube.com/watch?v=9XXgW34t40s" style="color:rgb(17,85,204);font-size:12.8000001907349px" target="_blank">https://www.youtube.com/watch?v=9XXgW34t40s</a></div><div><a href="https://www.youtube.com/watch?v=NLaDpGQuZDA" target="_blank">https://www.youtube.com/watch?v=NLaDpGQuZDA</a><br></div></div></div></div></div></div></div>
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