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Release Notes
Version 2.6.3
Document #: 50511
Release Notes Contents
Table of Contents
1 Introduction .......................................................................................................................... 7
List of Tables
Table 2-1: MP-20x Software Specifications................................................................................................................ 10
Notice
This document presents AudioCodes’ MP-20x Telephone Adapter Release Notes Version 2.6.3.
Information contained in this document is believed to be accurate and reliable at the time of printing.
However, due to ongoing product improvements and revisions, AudioCodes cannot guarantee accuracy of
printed material after the Date Published nor can it accept responsibility for errors or omissions. Updates to
this document and other documents can be viewed by registered customers at
www.audiocodes.com/support.
Tip: When viewing this manual on CD, Web site or on any other electronic copy, all cross-
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Trademarks
AC logo, Ardito, AudioCoded, AudioCodes, AudioCodes logo, CTI², CTI Squared, InTouch, IPmedia,
Mediant, MediaPack, MP-MLQ, NetCoder, Netrake, Nuera, Open Solutions Network, OSN, Stretto,
3GX, TrunkPack, VoicePacketizer, VoIPerfect, What's Inside Matters, Your Gateway To VoIP are
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Customer technical support and service are provided by AudioCodes’ Distributors, Partners, and
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Related Documentation
Document Title Document Number
MP-20x Telephone Adapter Quick Installation Guide LTRT-504xx (where xx refers to the document version)
MP-20x Telephone Adapter User's Manual LTRT-506xx
1 Introduction
The MP-20x is a 1 to 4 line (depending on model) Session Initiation Protocol (SIP) gateway, allowing
residential and small office / home office (SOHO) subscribers to connect ordinary “plain old telephone
service” (POTS) telephones or fax machines to the IP network. The MP-20x is interoperable with
leading softswitches and SIP Application Servers, offering legacy phone services such as caller ID, call
waiting, and call forwarding. In addition, the MP-20x includes an internal router with DHCP, NAT and
L2TP/PPTP/PPPoE capabilities, enabling subscribers to connect their home PC or LAN hub/switch to
the gateway.
Version 2.6.3 introduces a new MP-20x family member – the MP-203. MP-203 features two FXS lines
and an FXO line. When connected to the PSTN, the FXO line enables the user to make outgoing calls
to the PSTN and to receive incoming calls from the PSTN, in addition to the regular VoIP calls. The
MP-20x is suitable for users who wish to retain their PSTN line as a second line in addition to the VoIP
line.
Utilizing AudioCodes’ VoIPerfect™ core architecture, and gaining from its accumulated experience in
providing IP telephony solutions, the MP-20x series combines superior voice quality and state-of-the-art
features for end users, such as T.38 Fax Relay and G.168-2004-compliant Echo Cancellation. Low bit-
rate vocoders (voice coders) can be used simultaneously on both telephony ports to free valuable
bandwidth resources. The “Voice over Data” prioritization algorithm prevents degradation in voice
quality even during large data transfers.
The MP-20x series is designed for full interoperability with leading softswitches and SIP servers for
deployment in various network environments. Throughout the years, AudioCodes has invested
significant effort in establishing, and complying with, the leading and evolving VoIP standards. Support
of SIP, which is commonly found in Voice-over-Broadband (VoB) networks, assures seamless
integration and rapid deployment.
Reader's Notes
2 Version 2.6.3
2.1 What’s New in Version 2.6.3
Version 2.6.3 is the first Version for MP-203 Rev B models. MP-203 features two FXS ports and
one FXO port.
MP-203 FXO feature: The FXO interface enables users to connect a regular PSTN line to the MP-
203 and use it as a “second line” for incoming and outgoing calls. When making outgoing calls, the
user hears the VoIP dial tone by default. To access the PSTN, the user needs to dial a user-
defined DTMF key or sequence (using the “PSTN Access Code” parameter), for example, the key
‘9’ (which is the default access code). For all incoming calls (either from the VoIP or PSTN side),
the local phone will ring. In addition, the MP-203 user can receive waiting calls and perform call
transfer and three-way conferencing between VoIP and PSTN calls (for more information, refer to
the Addendum of the MP-20x User’s Manual).
Version 2.6.3 supports both MP-20x Rev A and Rev B models. The following products are
supported:
• MP-202/2FXS/SIP
• MP-201B/1FXS/SIP
• MP-202B/2FXS/SIP
• MP-203B/2FXS/1FXO/SIP
• MP-204B/4FXS/SIP
Warning: Do not download an MP-20x Rev B firmware to an MP-202 Rev A model and vice
versa. Typically, a protection mechanism prevents this and notifies the user of a
firmware mismatch. However, in certain cases (e.g. when downloading MP-20x
Rev A firmware to MP-20x Rev B models with 4-MB flash), this action can result in
an inoperable unit.
SIP proxy redundancy support: The Redundant Proxy feature enables the user to configure a
backup SIP proxy. Once this feature is enabled, MP-20x identifies cases where the primary proxy
does not respond to SIP signaling messages. In such a scenario, MP-20x registers to the
redundant proxy, and seamlessly continues normal functionality without the user noticing any
connectivity failure (i.e., non-traffic affecting).
Support for improved 'MGCP Like' digit map mechanism, as defined in the MGCP RFC 3435
Section 2.1.5. This new mechanism allows definition of more complex digit maps.
Re-answer (call regret) support. This feature enables the user to on-hook the phone during a call,
and then off-hook the phone again (i.e., regret on-hooking the call) within a user-defined timeout
(defined by the “Re-Answer Timeout” parameter). After the user picks up the phone receiver, the
previous call (conversion) can continue (i.e., the call is not disconnected). This feature is applicable
only when MP-20x is the called side (not the calling side).
Support for Do Not Disturb (DND) mode. The MP-202 model supports the DND feature according
to RFC 3326 and RFC 3261 Section 27.
Feature Details
Feature Details
outgoing call.
A silence period of about three seconds occurs after pressing the ‘Flash’ key during a conversation
(normally, the user presses ‘Flash’ + ’1’, ‘Flash’ + ’2’, or ‘Flash’ + ’3’). This limitation does not occur
when in ”Flash only” key sequence mode.
QoS traffic shaping: Enabling ‘TCP Serialization’ may cause problems viewing real-time video
streams on a PC that is connected to the device.
Caller ID Type II audio indication is sometimes heard by both the calling and the called parties. The
remote side doesn't hear the FSK; only an RFC 2833 DTMF tone.
To ensure that the OPTIONS Keep-Alive feature is fully functional, the remote side must send any
message, including the “501 Not implemented” SIP message.
Enabling or disabling the Periodic Checking of Configuration File feature requires a device reboot.
A SIP NOTIFY message with a “check-sync” event with a message body causes a device restart
even if there are ongoing conversation calls.
Reader’s Notes
A new configuration parameter – “Connect on 180” has been defined. When this parameter is
enabled, media is connected upon receipt of 180, 183 or 200 messages. When the parameter is
NOT enabled, media is connected upon receipt of 183 and 200 messages only.
MWI notification is NOT generated if there is an ongoing call. The indication is generated when the
call ends.
Added FaxMaxRate negotiation. Negotiation takes place for T.38 SDP media attribute
“T38MaxBitRate”. If the remote side requests a decrease in the value of this field, the DSP is
updated with this new value.
New regional settings supported for Argentina region.
The table below lists the MP-202 supported features:
OPTIONS Keep-Alive feature – To ensure this feature is fully functional, the remote side must send
any message, including the “501 Not implemented” message.
Periodic checking of configuration file - enabling or disabling this feature requires reboot.
SIP NOTIFY with a “check-sync” event with a message body causes system restart even if there
are ongoing conversation calls.
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