Hi Jeff, 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:
v=0 > o=user 0 0 IN IP4 162.212.130.252 > s=Session SIP/SDP > c=IN IP4 162.212.130.252 > t=0 0 > a=ice-ufrag:171m3 > a=ice-pwd:27g6nm2sol7btqvgper41odgjk > m=audio 55718 RTP/AVP 201 101 > a=rtpmap:201 OPUS/48000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=candidate:1 1 udp 2130706431 10.53.232.161 21000 typ host > a=candidate:3 1 udp 1694498815 188.29.165.133 49190 typ srflx raddr > 10.53.232.161 rport 21000 > a=candidate:2 1 udp 16777215 162.212.130.252 55718 typ relay raddr > 188.29.165.133 rport 49190 > a=candidate:1 2 udp 2130706430 10.53.232.161 21001 typ host > a=candidate:3 2 udp 1694498814 188.29.165.133 49191 typ srflx raddr > 10.53.232.161 rport 21001 > a=candidate:2 2 udp 16777214 162.212.130.252 57171 typ relay raddr > 188.29.165.133 rport 49191 On 8 December 2015 at 14:21, Jeff Pyle <[email protected]> wrote: > 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 >> >> > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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