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:
v=0. o=FreeSWITCH 1449573019 1449573020 IN IP4 192.168.5.5. s=FreeSWITCH. c=IN IP4 192.168.5.5. t=0 0. m=audio 11158 RTP/AVP 9 101. a=rtpmap:9 G722/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. >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. In order to make this data available for live calls, you'd probably have to store them in dialog variables. 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. - Jeff On Sat, Dec 5, 2015 at 11:21 PM, Nabeel <[email protected]> wrote: > Hello, > > 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? > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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