Telephony Interoperability Requirements

Prophecy uses SIP for call signaling. Once a session is established, it uses RTP for media processing between the SIP application server, the media server, and accompanying speech and TTS engines. This includes:

Accepting spoken input (inbound audio)
Playing synthesized speech and pre-recorded audio
Handling DTMF digits

Because Prophecy is a standards-based solution, it should work with any SIP device meeting the three standards described below.

Does Your Solution "Speak SIP?"

Your telephony must speak and understand SIP, as defined in RFC 3261 (Session Initiation Protocol). You can accomplish this directly through a SIP device or through an interface layer, such as a SIP gateway or a VoIP service.

To connect to TDM networks, Voxeo customers typically utilize standards-based SIP gateways from vendors such as Cisco and Sonus. These gateways convert traditional TDM circuits to VoIP when direct SIP trunking is not available from the carrier or switch.

Voxeo has performed extensive interoperability testing with SIP gateways from Cisco and uses these devices in its distributed hosted environment. Voxeo also offers a turnkey Prophecy IVR Server and Prophecy SIP Appliance, which include a built-in POTS or TDM-to-SIP gateway from Paraxip Technologies.

DTMF

Prophecy supports RFC 2833 for DTMF digits. Your telephony must send and receive out-of-band DTMF digits (i.e., where digits are passed via RTP packets). Now, this is relevant for a direct VOIP/SIP to Prophecy setup; however, in deployments where, for example, a Cisco 5400 is deployed with PRI trunks, it will be responsible for turning in-band tones delivered from the carrier to RFC 2833 and passing them to Prophecy.

By contrast, in-band DTMF is where DTMF digits are sent as tones; the media server determines whether the beep is of a frequency indicating a DTMF digit (a tone with meaning).

Prophecy does not currently support inband DTMF detection. However, it is planned for the near future.

Audio Codecs

Prophecy uses RTP for media processing between your telephony, the media server, and accompanying speech and TTS engines. Your telephony must support at least one of the following audio codecs:

G.711 ulaw
G.711 alaw
G.729 (This codec requires an additional licensing structure, but is supported by Prophecy.)
GSM (Global System for Mobile)

ANNOTATIONS: EXISTING POSTS

muks

6/10/2010 7:34 AM (EDT)

Can you please confirm whether Prophecy 9.0 supports inband DTMF generation?

VoxeoDustin

6/10/2010 12:49 PM (EDT)

Hello,

Prophecy supports DTMF via RFC2833, which is out-of-band DTMF. In order to process in-band DTMF, you will need a gateway that converts in-band DTMF to RFC2833.

However, if the SIP client does NOT negotiate a DTMF codec, then Prophecy will listen for in-band DTMF. You will need to ensure the SIP gateway or client does not offer any DTMF codecs for this to work properly. It is also not guaranteed to be as accurate as out-of-band DTMF since the DTMF tone will be in the media stream and is effected by a number of variables such as monotonocity, duration and quality of the tone.

Let me know if we can be of further assistance.

Regards,
Dustin Hayre
Solutions Engineer
Voxeo Corporation

Come join us at our Voxeo Customer Summit, June 21st - June 23rd at the Hard Rock Hotel.
Register today for your [url=http://www.voxeo.com/summits/customer]All Access Pass[/url].

muks

6/10/2010 8:47 PM (EDT)

Thanks guys for the quick response.

I didn't quite understand what you mean by "SIP client does NOT negotiate a DTMF codec" (sorry..still learning this tech). In our case the client is a mobile phone.

Assuming that "SIP client does NOT negotiate a DTMF codec", will Prophecy be able to process (detect and decode) the input given to it as inband DTMF? (without SIP gateway)

Does Prophecy support generation / sending of inband DTMF signals based on data provided(text) or we need a SIP gateway

Thanks
Muks

muks

6/10/2010 8:49 PM (EDT)

Sorry.. Could you also please suggest the SIP gateways to be used with Prophecy?

Thanks
Muks

VoxeoDustin

6/11/2010 1:59 PM (EDT)

Hello Muks,

Some form of SIP gateway will be necessary if you wish to send/receive calls from mobile or landlines. This could be a VOIP provider, a SIP trunk from a carrier, a SIP to analog/digital gateway, or an IP PBX.

You can find a list of support SIP gateways [url=https://evolution.voxeo.com/wiki/docs:prophecy_sip_gateway#supported_hardware]here[/url]. I highly recommend the Audiocodes devices for their ease of configuration and high availability features that the other devices do not have and they are much less expensive.

Let me know if you have any further questions.

Regards,
Dustin Hayre
Solutions Engineer
Voxeo Corporation

Come join us at our Voxeo Customer Summit, June 21st - June 23rd at the Hard Rock Hotel.
Register today for your [url=http://www.voxeo.com/summits/customer]All Access Pass[/url].

Setting Up Telephony

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