[OpenSIPS-Users] How to view bitrate, sampling rate and frame size during a SIP call?

[Jeff Pyle]

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 <nabeelshikder at gmail.com> 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 <jeff.pyle at fidelityvoice.com>
> 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 <nabeelshikder at gmail.com> 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?
>>>
>>
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