Support for DTMF (rfc4733, rfc2833)

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Mamadou

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Mar 26, 2013, 4:40:11 PM3/26/13

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Hello,

For information we have added support for DTMF in both webrtc2sip and sipml5.
As you already know, supports for DTMF in WebRTC is still in active discussion and there is no API yet. So, we decided to move forward like this:

- SIMPL5 193+
Now there is a keyPad button appearing in the 'options div' when the call is connected or early media (1xx with SDP) starts.
The client will always sends SIP INFO for DTMF which will work on any browser.

- webrtc2sip 64+
We have added a new configuration entry named 'dtmf-type' in config.xml (requires RTCWeb Breaker to be enabled). Possible values for this entry: 'rfc4733' (default) or 'rfc2833'.
'rfc4733' -> relay DTMF using RTP
'rfc2833' -> relay DTMF using SIP INFO
As an example, if 'dtmf-type' entry value is equals to 'rfc4733' then, any DTMF SIP INFO message from the browser to a SIP-legacy network will be sent as RTP-event packets as per RFC 4733.
More information in the technical guide: http://webrtc2sip.org/technical-guide-1.0.pdf

Regards,

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Mamadou DIOP - Technology Evangelist
Doubango Telecom - Paris, France
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navaismo

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Mar 26, 2013, 5:06:04 PM3/26/13

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Awesome Job! Thanks alot

Din Assegaf

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Mar 26, 2013, 8:44:57 PM3/26/13

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Thank you, we are waiting for this ..

-
Syeh Abidin

Capiez, Lionel

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Mar 27, 2013, 7:33:57 AM3/27/13

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Hi Mamadou,

Thanks a lot, this is great news !

I just gave it a try. DTMF as RTP payload works like a charm.

I would have the following feedbacks on ‘DTMF as SIP INFO’:

Content of SIP INFO message (with ‘Signal=’ and ‘Duration=’ fields) is that of ‘application/dtmf-relay’, but the Content-Type header would be missing to properly identify it as such (SIPML5 does sent it properly – webrtc2sip strips it off).

Why do you refer to it as rfc2833? My understanding is that rfc2833 is the ‘ancestor’ of rfc4733. They both define RTP payload for DTMF.

Although I have already encountered DTMF sent as SIP INFO with a Content-Type of ‘audio/telephone-event’ (defined in rfc2833/4733), in such a case, the content was identical to the binary content defined in those RFCs (here we get something that really is ‘application/dtmf-relay’, which, to my knowledge, is not defined in any RFC, but is common usage).

Best Regards,

Lionel

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Mamadou

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Mar 27, 2013, 12:34:53 PM3/27/13

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Starting webrtc2sip SVN r67+ the Content-Type header is added when relaying the INFO request.

Tayeb Meftah

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Mar 27, 2013, 12:38:00 PM3/27/13

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Lionel

True rfc2833 have nothing to do with sipinfo

Envoyé de mon iPhone

Capiez, Lionel

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Mar 27, 2013, 12:48:07 PM3/27/13

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Great.

Thanks for the quick fix.

Best Regards,

Lionel

BitesOne Lab

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Nov 4, 2013, 12:03:40 PM11/4/13

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Hi,

I have configured the WEBRTC2SIP GW to "rfc4733" but the SDP sent out in the INVITE to the PSTN GW does not offer the payload type of "telephone-event/8000" - as it should - to negotiate the use of it with the PSTN GW. Then, when SIP INFOs are sent by the client to the WEBRTC2SIP GW, the GW responds with 200 OK but nothing else happens.

Has anyone seen this problem?

Cheers / BO

Sumanta Sen

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Nov 6, 2013, 12:58:43 AM11/6/13

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Hi,

When you configure it as rfc4733, webrtc2sip will receive INFO but will not forward it as SIP INFO to other side. Instead, it will send it as in-band DTMF.

You are expected to get logs like this.

MSG: Sumanta: Function Find Transport exit()

*INFO: Processing INFO(dtmf-relay, Left->Right): Signal=2

Duration=120

*INFO: Gtw configured to relay DTMF using in-band(rfc4733) mode

*INFO: State machine: x0000_Connected_2_Connected_X_oDTMF

*INFO: Sending DTMF event...