I believe the absence of a ptime means a default of 20ms. That may be codec dependent; I don't recall.
- Jeff On Wed, Dec 9, 2015 at 1:11 AM, Nabeel <[email protected]> wrote: > 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
