<div dir="ltr">I believe the absence of a ptime means a default of 20ms. That may be codec dependent; I don't recall. <div><br></div><div><br></div><div>- Jeff</div><div><br></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Dec 9, 2015 at 1:11 AM, Nabeel <span dir="ltr"><<a href="mailto:nabeelshikder@gmail.com" target="_blank">nabeelshikder@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 dir="ltr">Hi Jeff,<div><br></div><div>Thanks for the information. I checked the SDPs, however mine does not have the 'a:ptime' line which could indicate the frame size. Is there a way to enable this? Here is an example of what I am seeing:</div><div><br></div><div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">v=0<br>o=user 0 0 IN IP4 162.212.130.252<br>s=Session SIP/SDP<br>c=IN IP4 162.212.130.252<br>t=0 0<br>a=ice-ufrag:171m3<br>a=ice-pwd:27g6nm2sol7btqvgper41odgjk<br>m=audio 55718 RTP/AVP 201 101<br>a=rtpmap:201 OPUS/48000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=candidate:1 1 udp 2130706431 10.53.232.161 21000 typ host<br>a=candidate:3 1 udp 1694498815 188.29.165.133 49190 typ srflx raddr 10.53.232.161 rport 21000<br>a=candidate:2 1 udp 16777215 162.212.130.252 55718 typ relay raddr 188.29.165.133 rport 49190<br>a=candidate:1 2 udp 2130706430 10.53.232.161 21001 typ host<br>a=candidate:3 2 udp 1694498814 188.29.165.133 49191 typ srflx raddr 10.53.232.161 rport 21001<br>a=candidate:2 2 udp 16777214 162.212.130.252 57171 typ relay raddr 188.29.165.133 rport 49191</blockquote><div> </div></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On 8 December 2015 at 14:21, Jeff Pyle <span dir="ltr"><<a href="mailto:jeff.pyle@fidelityvoice.com" target="_blank">jeff.pyle@fidelityvoice.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">OpenSIPS doesn't handle media so it has no knowledge of these things. You could glean some of this information by inspecting the offer and answer SDPs as they pass through. For example, here is an answer SDP that passed through reply_route attached to a 200 OK:<div><br></div><div><div>v=0.</div><div>o=FreeSWITCH 1449573019 1449573020 IN IP4 192.168.5.5.</div><div>s=FreeSWITCH.</div><div>c=IN IP4 192.168.5.5.</div><div>t=0 0.</div><div>m=audio 11158 RTP/AVP 9 101.</div><div>a=rtpmap:9 G722/8000.</div><div>a=rtpmap:101 telephone-event/8000.</div><div>a=fmtp:101 0-16.</div><div>a=ptime:20.</div></div><div><br></div><div>From this data you know it's 8khz sampling rate, since it's G722 you know it's a 64kbps bitrate, and the ptime is 20ms. You'd have to account for future in-dialog requests (reINVITEs and UPDATEs) that may change these parameters.</div><div><br></div><div>In order to make this data available for live calls, you'd probably have to store them in dialog variables.</div><div><br></div><div>In other words, it may be possible to maintain this data from within OpenSIPS, but it becomes complicated quickly depending on the variety of endpoints and applications you use. It is generally easier to gather this data from the endpoints themselves but you've already said your app does not have a way to do that. That's unfortunate.</div><div><br></div><div><br></div><div>- Jeff</div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote"><span>On Sat, Dec 5, 2015 at 11:21 PM, Nabeel <span dir="ltr"><<a href="mailto:nabeelshikder@gmail.com" target="_blank">nabeelshikder@gmail.com</a>></span> wrote:<br></span><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span><div dir="ltr">Hello,<div><br></div><div>I need to view the active sampling rate, bitrate and frame size during a SIP call. The app currently does not have a user interface or custom function to display this. Is there any other way I can view these parameters during a live call? What is the simplest way to do this?</div></div>
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