Sorry. I was asking the question in more general sense about RTSP clients interpreting the bandwidth parameter of the SDP. <div><br></div><div>The other question of controlling streaming framerate, where can I modify this parameter ?</div>
<div><br></div><div><br><div class="gmail_quote">On Tue, Nov 29, 2011 at 7:30 PM, Ross Finlayson <span dir="ltr"><<a href="mailto:finlayson@live555.com">finlayson@live555.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"><div><div class="im"><blockquote type="cite"><div>On Android clients RTSP playback has a latency of around 5-6 seconds. The client buffers data for this duration and then starts playing. </div>
<div>For live streaming this latency is very high. </div>
<div><br></div><div><span style="font-family:Arial,'Liberation Sans','DejaVu Sans',sans-serif;font-size:14px;line-height:18px;background-color:rgb(255,255,255)">Does the client use the bandwidth information (b=AS:<bandwidth>) to calculate the buffer size. e.g 5-6sec in this case? </span></div>
</blockquote><div><br></div></div>How would we know? You'll have to ask whoever developed the client.</div><span class="HOEnZb"><font color="#888888"><br><div>
<span style="border-collapse:separate;color:rgb(0,0,0);font-family:Helvetica;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:-webkit-auto;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;font-size:medium"><span style="border-collapse:separate;color:rgb(0,0,0);font-family:Helvetica;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:-webkit-auto;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;font-size:medium">Ross Finlayson<br>
Live Networks, Inc.<br><a href="http://www.live555.com/" target="_blank">http://www.live555.com/</a></span></span>
</div>
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<br></blockquote></div><br></div>