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Add a command that will leave only one telephone-event line in SDP

Aperta
  • Qualcosa non va?
#2,044 0 commenti 0 reazioni 0 assegnatari Vedi su GitHub

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Valutazione

Difficoltà
4/5
Tempo stimato
3-5 giorni
Idoneità per principianti
20/100
Tipo di issue
Funzionalità
Chiarezza
Abbastanza chiara
Stato di attività
Ferma
Ambito
audio-video-rtc

Direzione di ricerca

Start from how rtpengine_offer flags are parsed, then trace where the outgoing SDP is assembled and where codec-transcode appends the telephone-event/48000 rtpmap alongside the OPUS and PCMA entries. The two SDP exchanges in the report (offer to the client and the answer that ends up with no telephone-event) are the test case. Done means a flag, or a documented flag combination, that leaves only one telephone-event line when transcoding PCMA to OPUS. The report has gone unanswered for months and no approach has been agreed.

Scritto dal modello di indicizzazione a partire dal testo della issue.

Descrizione

Is your feature request related to a problem? Please describe

Hello,
The problem is that there are some SIP clients (Mac telephone, Zoiper, Microsip 3.20) that, if they are received 2 telephone-events in SDP for OPUS and for the 711 codec, select the wrong one in SDP and cannot use DTMF afterwards. The following flags are used when calling rtpengine_offer: rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP replace-origin codec-transcode-opus codec-transcode-PCMA codec-transcode-telephone-event, so telephone-event/48000 is always added.
Question: is there any way to solve this with the correct rtpengine flags, or just get rid of buggy clients?
Or maybe it's possible to add a flag that won't add telephone-event/48000 to every SDP when we want transcoding between PCMA and OPUS?

Examples:
SDP in INVITE to the client:

m=audio 11492 RTP/AVP 8 96 97 101
c=IN IP4 Y.Y.Y.Y
a=rtpmap:8 PCMA/8000
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 telephone-event/48000
a=fmtp:97 0-15
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:11493
a=ptime:20
a=maxptime:140

Client's answer:

c=IN IP4 X.X.X.X
t=0 0
m=audio 58085 RTP/AVP 8 0 97
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=sendrecv
a=rtcp-mux

And result SDP after rtpengine process answer - it goes without telephone-event because SIP-client used wrong dtmf, 97 instead 101:

m=audio 11874 RTP/AVP 8
c=IN IP4 X.X.X.X
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtcp:11875
a=ptime:20
Describe the solution you'd like

Maybe it's possible to add a flag that won't add telephone-event/48000 to every SDP when we want transcoding between PCMA and OPUS?

Describe alternatives you've considered

No response

The rtpengine version you checked that didn't have the feature you are asking for

13.5.1.2+0~mr13.5.1.2 git-mr13.5.1-876b260

Lingua principale
Perl
Stelle
992
Fork
459
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