<div dir="ltr"><span style="font-size:13px">"I tried adding this line to vars.xml: </span><span style="font-size:13px"><font face="sans-serif"><span style="line-height:19.2000007629395px">(<param name="renegotiate-codec-on-reinvite" value="true”/>"</span></font></span><div><font face="sans-serif"><span style="line-height:19.2000007629395px"><br></span></font></div><div><font face="sans-serif"><span style="line-height:19.2000007629395px">This would be because its not a variable, its a sofia profile param that you would add to a profile thats in the sip_profiles directory.<br></span></font><div><span style="font-size:13px"><font face="sans-serif"><span style="line-height:19.2000007629395px"><br></span></font></span></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Jan 13, 2015 at 11:01 PM, Jon Lederman <span dir="ltr"><<a href="mailto:jonlederman@gmail.com" target="_blank">jonlederman@gmail.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">Hi,<div><br></div><div>I am trying to get my client to work with Freeswitch in the following context to no avail. If I initiate a video call, the video works fine. However, if I start an audio call and then try to add video later in the call, no video is received. I tried adding this line to vars.xml: <span style="background-color:rgb(255,255,255)"><font face="sans-serif"><span style="line-height:19.200000762939453px">(<param name="renegotiate-codec-on-reinvite" value="true”/></span></font></span></div><div><span style="font-family:sans-serif;font-size:13px;line-height:19.200000762939453px;background-color:rgb(255,255,255)">That, </span><font face="sans-serif"><span style="line-height:19.200000762939453px">however, did not fix the problem.</span></font></div><div><font face="sans-serif"><span style="line-height:19.200000762939453px"><br></span></font></div><div><font face="sans-serif"><span style="line-height:19.200000762939453px">Any help would be greatly appreciated. The SDP negotiation is indicated below. Thanks in advance.</span></font></div><div><br></div><div>I do see in fs_cli that FS has received a new SDP and is trying to renegotiate as follows:</div><div><br></div><div>INITIAL SDP (AUDIO ONLY):</div><div><div style="font-family:-apple-system-font;line-height:16px"><div>v=0</div><div>o=FreeSWITCH 1421171138 1421171140 IN IP4 146.148.54.28</div><div>s=FreeSWITCH</div><div>c=IN IP4 146.148.54.28</div><div>t=0 0</div><div>m=audio 25746 RTP/AVP 124 101</div><div>a=rtpmap:124 opus/48000/2</div><div>a=fmtp:124 useinbandfec=1</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=silenceSupp:off - - - -</div><div>a=ptime:20</div><div>a=sendrecv</div><div>m=video 26852 RTP/AVP 98</div><div>a=rtpmap:98 VP8/90000</div></div><div style="font-family:-apple-system-font;line-height:16px"><br></div><div style="font-family:-apple-system-font;line-height:16px">NEW SDP WITH ATTEMPT TO ADD VIDEO TO EXISTING SESSIOn:</div><div style="font-family:-apple-system-font;line-height:16px"><div>2015-01-14 00:55:29.401667 [DEBUG] sofia.c:5828 Remote SDP:</div><div>v=0</div><div>o=1000 3968 2153 IN IP4 10.1.10.10</div><div>s=Talk</div><div>c=IN IP4 10.1.10.10</div><div>b=AS:380</div><div>t=0 0</div><div>a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics</div><div>m=audio 53368 RTP/AVP 124 101</div><div>a=rtpmap:124 opus/48000/2</div><div>a=fmtp:124 useinbandfec=1; stereo=0; sprop-stereo=0</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-15</div><div>m=video 9078 RTP/AVP 103</div><div>a=rtpmap:103 VP8/90000</div></div></div><div><br></div><div><br></div></div><br>_________________________________________________________________________<br>
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<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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