the feature can be found on Cisco.com at www.cisco.com/go/cube, "Configure," "Configuration Examples and TechNotes." Media The speech path, or Real-Time Transport Protocol (RTP) stream, of an IP call typically flows directly between the communicating endpoints, requiring that the endpoints sup- port a common set of capabilities for voice or video encoding. These streams can be encoded in myriad ways and still be standards-compliant, and it is likely that your enterprise endpoints support only a subset of these methods, or that you want to deploy only a subset of them to ensure that the interconnect policies of your net- work are met. Media interworking areas to consider as your network migrates to end-to-end IP, or start to use SIP trunking for PSTN access, include DTMF digits Codecs, including transcoding and transrating Fax and modem traffic DTMF digits can be transported in numerous ways in both H.323 and SIP. How DTMF digits are exchanged between endpoints and applications often requires conversion or translation between these methods. DTMF Relay For G.711 streams DTMF digits can reasonably be carried in-band as tones within the speech path itself, but for low bit-rate codecs such as G.729 and Internet Low-Bitrate Codec (iLBC) the encoding algorithm distorts the DTMF frequencies to the point that Chapter 8: Interworking 149 DTMF recognizing success rates with voice mail and interactive voice response (IVR) sys- tems become unacceptably low. For this reason DTMF relay methods are predominant in Voice over IP (VoIP) streams. DTMF relay means that the DTMF tones are encoded in a manner different from voice sample encoding and are re-created as clear tones at the destination endpoint by its local Digital Signal Processor (DSP). That way, network transport does not distort the frequen- cies of the tones. Some DTMF relay methods encode just the tone pressed by the origi- nator; others also encode the duration of the tone, enabling higher accuracy in play-out at the destination endpoint. Although a handful of SIP tr