Change SipPlugin's dtmf pt hardcoded to get from sdp

[ywmoyue]

Fix bug raised by @ycherniavskyi from #3280 (comment)

#3280 set a hardcoded payload type of 101 for the telephone-event RTP payload format.

This PR Change dtmf payload hardcoded to get the telephone-event payload value from sdp

[lminiero]

Thanks for the fix! I've added a couple of notes inline.

As a side note, while doing some unrelated tests today, I noticed that your DTMF detection only seems to be working when using "master" sessions, and not when using "helper" sessions instead (that is, the additional call lines you can associate to the main registration). I didn't have time to investigate yet, but I just thought I'd raise it to your attention, in case you have cycles to debug that too.

[ywmoyue]

First of all, thank you for notes, and secondly, thank's for the reminder about the 'helper' session. I haven't used 'helper' session before, I will use and test this dtmf detection later.

For the first note，did you mean the janus_sip_media_reset function? I have just modified it. I am not sure whether I need to add the dtmf_pt field to janus_no_sip_media struct.

[lminiero]

did you mean the janus_sip_media_reset function?

Yes, apologies for the typo!

[lminiero]

I haven't used 'helper' session before, I will use and test this dtmf detection later.

It may also be a bug in the demos, as that's what I was using to test a call between two users via the SIP plugin and injecting DTMF digits. I'll make some more tests on that later.

[lminiero]

It may also be a bug in the demos

That's indeed it: there's a bug when using the DTMF controls from the SIP demo on helper calls. If I manually invoke the DTMF sending methods, DTMF tones are correctly intercepted on the other end, and correctly sent as events on the right helper handle. I'll fix that demo bug separately.

[lminiero]

Looks good to me, merging then!