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Negotiation of the optimal voice codec is sometimes dependent on the PHONE ADAPTER device’s
ability to “match” a codec name with the far-end device/gateway codec name. The PHONE
ADAPTER allows the network administrator to individually name the various codecs that are
supported such that the correct codec successfully negotiates with the far end the equipment.
7.2.3. Secure Calls
A user (if enabled by service provider or administrator) has the option to make an outbound call
secure in the sense that the audio packets in both directions are encrypted.
7.2.4. Voice Algorithms:
7.2.4.1. G.711 (A-law and mµ-law)
This very low complexity codec supports uncompressed 64 kbps digitized voice transmission at one
through ten 5 ms voice frames per packet. This codec provides the highest voice quality and uses the
most bandwidth of any of the available codecs.
7.2.4.2. G.726
This low complexity codec supports compressed 16, 24, 32 and 40 kbps digitized voice transmission
at one through ten 10 ms voice frames per packet. This codec provides the high voice quality.
7.2.4.3. G.729A
The ITU G.729 voice coding algorithm is used to compress digitized speech. Linksys supports
G.729. G.729A is a reduced complexity version of G.729. It requires about half the processing power
to code G.729. The G.729 and G.729A bit streams are compatible and interoperable, but not
identical.
7.2.4.4. G.723.1
The PHONE ADAPTER supports the use of ITU G.723.1 audio codec at 6.4 kbps. Up to 2 channels
of G.723.1 can be used simultaneously. For example, Line 1 and Line 2 can be using G.723.1
simultaneously, or Line 1 or Line 2 can initiate a 3-way conference with both call legs using G.723.1.
7.2.5. Codec Selection
The administrator can select which low-bit-rate codec to be used for each line. G711a and G711u
are always enabled.
7.2.6. Dynamic Payload
When no static payload value is assigned per RFC 1890, the PHONE ADAPTER can support
dynamic payloads for G.726.
7.2.7. Adjustable Audio Frames Per Packet
This feature allows the user to set the number of audio frames contained in one RTP packet. Packets
can be adjusted to contain from 1 – 10 audio frames. Increasing the number of packets decreases the
bandwidth utilized – but it also increases delay and may affect voice quality.
7.2.8. Fax Tone Detection Pass-Through
Users can connect a fax terminal to the PHONE ADAPTER telephone port(s). Fax terminals transmit
a single tone when they answer a call. The PHONE ADAPTER detects the type of equipment in use
on the basis of its answer tone. When it detects the equipment answering the call, the PHONE
ADAPTER performs a switchover from the current audio codec to G.711 codec.
7.2.9. DTMF: In-band & Out-of-Band (RFC 2833) (SIP INFO *)
© 2004 Linksys Proprietary (See Copyright Notice on Page 2)
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