Telos Zephyr Xstream User Manual page 237

Ethernet (SIP) and Ethernet (RTP)
This selection enables Zephyr Xstream's built‐in Ethernet interface for audio
coding and/or decoding. This allows for transmission and reception of audio
over Internet Protocol (IP) based networks.
The (SIP) and (RTP) designations refer to what protocol is used to establish
communication. SIP (Session Initiation Protocol) connections are bi‐directional.
By 'dialing' another Zephyr Xstream, you request a return feed as well.
RTP (Real‐time Transport Protocol) is a mono‐directional, 'push only' mode.
When you 'dial' another Zephyr Xstream in RTP mode, you send audio to the
remote site, but receive nothing in return unless the remote operator manually
initiates a connection back to the studio.
Appendix 6 covers this topic in detail.
w
IMPORTANT!
The following menus will have different options when the Interface is set to Ethernet.
• CODEC
• DIAL
Prefix:
This number is appended to the beginning of every number dialed. Useful when a "9" or other
digit is required for access.
t
DEEP TECH NOTE!
Long Distance services that rely on "in-band" tones (i.e. DTMF) to convey password
information to the service provider cannot be accessed on Circuit Switched Data (Call type =
Zephyr) data calls.
This is because there is no audio path to send tones. Only if the network had an MPEG codec,
would it be able to "hear" the tones.
The entire call setup string is sent as digital information on the D channel.
USER'S MANUAL
Section 11: THE WORKS – Detailed Menu Reference 225