dtmf inbound to rfc2833

[song gao]

Hi, 
    my rtpengine version is : 13.1.0.0+0~mr13.1.0.0 git-heads/mr13.0.1.2-3ac08574 
    I want to change  dtmf inbound to rfc2833,  the incoming invite sdp include : 101 telephone-event/8000. 

 I tested  flags " always-transcode detect-DTMF codec-transcode-telephone-event codec-strip=telephone-event  " or "always-transcode inject-DTMF codec-transcode-telephone-event codec-strip=telephone-event , they are not work. the in-band tones stay inbound.

 Please  give some suggestions

   

 Thanks.

[Richard Fuchs]

Just adding `detect-DTMF` should be enough.

Cheers

[song gao]

hi rfuchs,

   Thanks for your quick reply, I followed your advice and only used detect-DTMF, still not work.

  Do I need to upgrade the rtpengine version to the latest version? 
here is my offer:
```
[1749623982.488811] DEBUG: [5b4c6cc1a4b64149921298fbc94bb25f]: [control] Dump for 'offer' from 172.16.4.111:36555: { "sdp": "v=0

o=- 3958641637 3958641637 IN IP4 172.16.80.13

s=pjmedia

b=AS:84

t=0 0

a=X-nat:0

m=audio 4000 RTP/AVP 0 8 101

c=IN IP4 172.16.80.13

b=TIAS:64000

a=rtcp:4001 IN IP4 172.16.80.13

a=sendrecv

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ssrc:2086618623 cname:3dab6d0078116da1

", "media-address": "172.16.4.111", "DTLS": "off", "ICE": "remove", "flags": [ "detect-DTMF", "SDES-off" ], "replace": [ "origin" ], "transport-protocol": "RTP/AVP", "call-id": "5b4c6cc1a4b64149921298fbc94bb25f", "received-from": [ "IP4", "172.16.80.13" ], "from-tag": "1016626a4c374841b870e6acaf261d50", "command": "offer" }

```

[Richard Fuchs]

It's always worth trying with the newest version.

You can also experiment with the new codec-ignore option.

Cheers

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[song gao]

 Hi,

    I had already update rtpengine  version to `Version: 13.3.1.7+0~mr13.3.1.7 git-HEAD-c2223681`, `detect-DTMF` is not work. 

    Here is opensips send `invite` ,  rtpengine_manager flag: codec-accept=all detect-DTMF codec-ignore-telephone-event DTLS=off SDES-off ICE=remove RTP/AVP

   1. here is my microPhone send invite sdp:

   ```

v=0

o=- 3959747936 3959747936 IN IP4 172.16.80.13

s=pjmedia
b=AS:84
t=0 0
a=X-nat:0

m=audio 4002 RTP/AVP 0 8 101

c=IN IP4 172.16.80.13
b=TIAS:64000

a=rtcp:4003 IN IP4 172.16.80.13

a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

a=ssrc:765414956 cname:69094e7b13201f35

   ```

    2. opensips  forward the `invite` sdp:

```

v=0
o=- 3959747936 3959747936 IN IP4 172.16.4.111

s=pjmedia
b=AS:84
t=0 0
a=X-nat:0

m=audio 30048 RTP/AVP 0 8 101
c=IN IP4 172.16.4.111
b=TIAS:64000

a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

a=ssrc:765414956 cname:69094e7b13201f35
a=sendrecv
a=rtcp:30049

```

 3. opensips receive `183` sdp:

```

v=0
o=UAP9600 294609 294609 IN IP4 172.16.7.208
s=Sip Call
c=IN IP4 172.16.7.209
t=0 0
m=audio 10100 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

a=ptime:20
a=fmtp:101 0-15
a=sendrecv

```

4. opensips forward `183` to microphone sdp:

```

v=0
o=UAP9600 294609 294609 IN IP4 172.16.4.111
s=Sip Call
t=0 0
m=audio 30122 RTP/AVP 8 0
c=IN IP4 172.16.4.111
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtcp:30123
a=ptime:20

```

   Could you please provide a complete and useful command parameter?