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MP-20x Telephone Adapter

Release Notes
Version 2.6.3
Document #: 50511
Release Notes Contents

Table of Contents

1  Introduction .......................................................................................................................... 7 

2  Version 2.6.3 ........................................................................................................................ 9 


2.1  What’s New in Version 2.6.3 ...................................................................................................... 9 
2.2  Resolved Constraints in Version 2.6.3 ..................................................................................... 11 
2.3  Known Limitations in Version 2.6.3 .......................................................................................... 12 
3  Previous Version 2.6.2 ...................................................................................................... 13 
3.1  What’s New in Version 2.6.2 .................................................................................................... 13 
3.2  Resolved Constraints in Version 2.6.2 ..................................................................................... 13 
3.3  Known Limitations in Version 2.6.2 .......................................................................................... 13 
4  Previous Version 2.6.1 ...................................................................................................... 15 
4.1  What’s New in Version 2.6.1 .................................................................................................... 15 
4.2  Resolved Constraints in Version 2.6.1 ..................................................................................... 16 
4.3  Known Limitations in Version 2.6.1 .......................................................................................... 17 
5  Previous Version 2.6.0 ...................................................................................................... 19 
5.1  What’s New in Version 2.6.0 .................................................................................................... 19 
5.2  Resolved Constraints in Version 2.6.0 ..................................................................................... 20 
5.3  Known Limitations in Version 2.6.0 .......................................................................................... 20 

Version 2.6.3 3 November 2008


MP-20x

List of Tables
Table 2-1: MP-20x Software Specifications................................................................................................................ 10 

Release Notes 4 Document #: LTRT-50511


Release Notes Notices

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.

© Copyright 2008 AudioCodes Ltd. All rights reserved.


This document is subject to change without notice.
Refer to any current documentation that may be included with your hardware delivery.
Date Published: November-25-2008

Tip: When viewing this manual on CD, Web site or on any other electronic copy, all cross-
references are hyperlinked. Click on the page or section numbers (shown in blue) to
reach the individual cross-referenced item directly. To return to the point from where
you accessed the cross-reference, press Alt + Å.

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
trademarks or registered trademarks of AudioCodes Limited.
All other products or trademarks are the property of their respective owners.

WEEE EU Directive
Pursuant to the WEEE EU Directive, electronic and electrical waste must not be disposed of with
unsorted waste. Please contact your local recycling authority for disposal of this product.

Customer Support
Customer technical support and service are provided by AudioCodes’ Distributors, Partners, and
Resellers from whom the product was purchased. For Customer support for products purchased directly
from AudioCodes, contact support@audiocodes.com.

Abbreviations and Conventions


Each abbreviation, unless widely used, is spelled out in full when first used, and only industry-standard
terms are used throughout this manual. 0x before a number denotes hexadecimal notation. DSP (Digital
Signaling Processor) and VoPP (Voice over Packet Processor) may be used interchangeably.

Version 2.6.3 5 November 2008


MP-20x

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

Release Notes 6 Document #: LTRT-50511


Release Notes 1. Introduction

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.

Version 2.6.3 7 November 2008


MP-20x

Reader's Notes

Release Notes 8 Document #: LTRT-50511


Release Notes 2. Version 2.6.3

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.

Version 2.6.3 9 November 2008


MP-20x

The MP-20x software specifications are summarized in the following table:

Table 2-1: MP-20x Software Specifications

Feature Details

VoIP Signaling Protocols ƒ SIP - RFC 3261, RFC 2327 (SDP)


Data Protocols ƒ IPv4, TCP, UDP, ICMP, ARP,TLS (SIP Over TLS)
ƒ PPPoE (RFC 2516)
ƒ L2TP (RFC 2661)
ƒ PPTP (RFC 2637)
ƒ DNS, Dynamic DNS
ƒ WAN–to-LAN Layer 3 routing with:
9 DHCP Client/Server (RFC 2132)
9 NAT: RFC 3022, Application Layer Gateway (ALG)
9 Stateful Packet Inspection Firewall
9 QoS - Priority queues, VLAN 802.1p,Q tagging, traffic shaping
or
Layer 2 switching (currently not supported)
ƒ STUN (RFC 3489)
Media Processing ƒ Voice Coders: G.711, G.723.1, G.729A/B, G.726
Optional - iLBC, AMR (separate software image)
ƒ Echo Cancelation: G.168-2004 compliant, 64-msec tail length
ƒ Silence Compression
ƒ Adaptive Jitter Buffer 300 msec
ƒ Fax bypass, Voice-Band Data and T.38 fax relay
ƒ Automatic Gain Control
Telephony Features ƒ Call Hold and Transfer
ƒ Call Waiting
ƒ 3-Way Conferencing (currently, not supported on MP-204)
ƒ Message Waiting Indication
ƒ Call Forward
Configuration/ Management ƒ Embedded Web Server for configuration and management
ƒ TR-069 and TR-104 for remote configuration and management
ƒ Remote firmware upgrade and configuration by HTTP,TFTP,FTP and HTTPS
ƒ Configuration file encryption (3DES)
ƒ SIP-triggered remote firmware and configuration upgrade
ƒ Command-Line Interface (CLI) over Telnet
ƒ Dual image management
Packetization ƒ RTP/RTCP Packetization (RFC 3550, RFC 3551)
ƒ DTMF Relay (RFC 2833)
Security ƒ HTTPs for Web-based configuration
ƒ Password protected Web pages (MD5)

Release Notes 10 Document #: LTRT-50511


Release Notes 2. Version 2.6.3

Feature Details

Telephony Signaling ƒ In-band:


9 DTMF: Detection and Generation, TIA464B
9 Caller ID: Telcordia, ETSI, NTT - Type I, Telcordia Type II
9 Call Progress Tones
ƒ Out-of-band:
9 FXS Loop-start Signaling
9 On/Off Hook, Flash Hook

2.2 Resolved Constraints in Version 2.6.3


The following bugs were resolved in Version 2.6.3:
„ Three-way conferencing with local mixing is now supported on MP-204 Rev B models.
„ Changing the G.726/16 Payload Type twice resulted in voice not being audible on both sides; now,
voice is heard.
„ Caller ETSI DTMF CID is now supported (for Portugal).
„ The MP-20x Rev B model now supports Polarity Reversal.
„ The limitation of the maximum number of dialed digits has now been increased to 1,024.
„ MP-20x can now be configured to send FSK Caller ID before ringing (using a new Caller ID
parameter - “Prior to Ring”).
„ MP-20x now responds with Busy when a fax call is received during a fax session.
„ MP-202 now answers upon receiving a SIP 407 after sending a SIP Re-INVITE over TCP.
„ T.38 fax transmission is now functioning when the remote fax machine is Super G3.
„ The Time Description line is now included in SIP OK messages upon a VBD Re-INVITE from
remote machines.
„ Regional settings has now been added for Portugal.
„ MP-20x can now be configured to use different DSCP values for RTP and SIP.

Version 2.6.3 11 November 2008


MP-20x

2.3 Known Limitations in Version 2.6.3


Version 2.6.3 includes the following known limitations:
„ MP-203B/2FXS/1FXO includes the following known limitations:
• The first configured VoIP codec is also used internally for calls between the local phone and
the PSTN. If a codec other than G.711 is configured as the first codec, then voice quality
degradation occurs in PSTN calls.
• DTMF transport over SIP should not be selected.
• FXO ring detection does not support all possible ring cadences.
• When the user answers a waiting PSTN call during a VoIP call, it is not possible to return to
the VoIP call.
„ Faxes can only be sent between two local lines if the selected fax transport mode is Transparent
(in G.711).
„ The Web interface is not automatically refreshed during the firmware upgrade process.
„ After the dial tone timeout has expired (and a fast-busy tone is played), the user can still make an
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.

Release Notes 12 Document #: LTRT-50511


Release Notes 3. Previous Version 2.6.2

3 Previous Version 2.6.2


3.1 What’s New in Version 2.6.2
„ Version 2.6.2 is the first General Availability (GA) Version of MP-20x Rev B and is based on
Version 2.6.1. The following three MP-20x Rev B models are supported in this version:
• MP-201 Rev B, providing one FXS interface
• MP-202 Rev B, providing two FXS interfaces
• MP-204 Rev B, providing four FXS interfaces
„ Version 2.6.2 supports only MP-20x Rev B devices. The next version – 2.6.3 – will be the first GA
version to support both Rev A and Rev B devices.
„ Dual firmware image support on 8-MB flash devices: MP-20x Rev B models provide an increased
flash memory of 8 MB, which is capable of storing two firmware images (the primary image and a
recovery image, described below). For previous MP-20x devices, which provide only 4 MB of flash
memory, a separate Version 2.6.2 firmware file is available.
„ Recovery firmware support: In addition to the primary firmware, the MP-20x stores a recovery
firmware, which is executed only if the primary image is missing or damaged (e.g. if the user
unplugs the power during firmware upgrade). Except for the analog or VoIP interfaces, the
recovery image supports all other interfaces and enables the MP-20x to reconnect to the Internet
and download the primary firmware.

3.2 Resolved Constraints in Version 2.6.2


The following bugs were resolved in Version 2.6.2:
„ When pressing ‘Flash’ + ’1’ or ‘Flash’ + ’2’ (for call hold and transfer), the DTMF signal is
sometimes audible at the remote side.
„ When the key sequence is set to “flash + digits sequence”, only one-way voice occurs in blind
transfer.
„ When the key sequence is set to “flash + digits sequence”, a warning tone is heard after transfer
and semi-attended transfer.
„ When the key sequence is set to “flash + digits sequence”, a warning tone is heard after returning
to the first call when the second call is not answered.
„ When the key sequence is set to “flash + digits sequence”, a warning tone is not heard upon an
unanswered call.

3.3 Known Limitations in Version 2.6.2


Version 2.6.2 includes the following known limitations:
„ Three-way conferencing with local mixing is currently not supported on MP-204 Rev B devices.
This feature will be added in Version 2.6.2 for some hardware versions.
„ Faxes can only be sent between two local lines if the selected fax transport mode is Transparent
(in G.711).
„ The Web interface is not automatically refreshed during the firmware upgrade process.
„ After the dial tone timeout has expired (and a fast-busy tone is played), the user can still make an

Version 2.6.3 13 November 2008


MP-20x

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.

Release Notes 14 Document #: LTRT-50511


Release Notes 4. Previous Version 2.6.1

4 Previous Version 2.6.1


4.1 What’s New in Version 2.6.1
The following new features are supported in Version 2.6.1:
„ The MP-202 now supports configuration file encryption. Encrypted files include the extension .cfx
(instead of .cfg) or .inx (instead of .ini). The MP-202 automatically identifies the .cfx or .inx
extensions and tries to decrypt the file.
To encrypt a configuration file, run the following command (on Linux or Windows):
openssl des3 -in <original file> -out <encrypted file> -k <password> -S
<salt value>
Where,
• <original file> = original clear text configuration file (.cfg or .ini)
• <encrypted file> = an encrypted output file (.cfx or .inx)
• <password> = password used to encrypt the file
• <salt value> = 8 bytes of a special key value that is combined with the password. The format
consists of 16 hexadecimal digits [0-9,A-F].
For example:
openssl des3 -in c:\temp\try_enc_conf.cfg -out c:\temp\try_enc_conf.cfx
-k MyPassword123456 -S 0123456789ABCDEF
„ The MP-202 now supports internet Low Bit Rate Codec (iLBC).
„ The MP-202 now enables configuration of 40-ms and 60-ms packetization time for codec G.729.
„ TR-069 interoperability enhancements with the Dimark (Friendly) ACS.
„ The MP-202 now enables configuration of the TCP Session TTL, using the new parameter 'TCP
Session Timeout'. The default value is one hour. This parameter can be configured in the 'Security
- General' screen.
„ The MP-202 now supports the Brazilian Portuguese language.
„ The MP-202 now enables disabling the password's encryption in the configuration file. The .cfg or
.ini files that are downloaded from a remote server can now include textual values (e.g. passwords)
that are automatically hidden before burning to flash. To indicate that a value must be hidden, use
{“value”} (instead of just value).
Below are some examples of this feature:
• .ini file: rg_conf/voip/line/1/auth_password={"foobaa"}
• .cfg file: (auth_password({"foobaa"}))
„ The MP-202 now supports the following new configurable fax parameters:
• ImageDataRedundancy:Redundancy: Level for output Image Data (2400…14400 bps).
♦ 0 = No redundancy.
♦ 1-3 = Redundancy level.
• T30ControlDataRedundancy: Redundancy level for output T.30 Control Data (300 bps).
♦ 0 = No redundancy.
♦ 1-7 = Redundancy level.

Version 2.6.3 15 November 2008


MP-20x

• FaxModemJitter: Fax Relay Jitter Buffer configuration.


♦ 0 = Adaptive Jitter Buffer. The AC49x device sets the Jitter Buffer size automatically and
then adapts it according to network conditions.
♦ 1-511 = Fixed Jitter Buffer size (in msec).
„ When configuring the key sequence style to "Send Flash Hook via SIP", the user can modify the
INFO message that is sent upon Flash. The user can change the following fields:
• Content Type header field
• Message Body field
For example:
• SIP INFO Header = application/broadsoft; version = 1.0
- or / and -
• SIP INFO Body = event flashhook
The update is performed in the 'Voice Over IP - Dialing' page, under the 'Key Sequence' group.
„ Disables the call waiting tone when in Fax mode.

4.2 Resolved Constraints in Version 2.6.1


The following bugs were fixed in Version 2.6.1:
„ MP-202 does not send a second REFER message after the first one is replied with a '401 Not
Authorized' SIP response.
„ Handling of re-INVITE messages cause memory 'leakage' – this was fixed by disabling the Session
version condition check.
„ MP-202 always sends 14400 as the bit rate in T.38 SIP messages.
„ MP-202 doesn't update the DSP after RFC 2833 SIP negotiation.
„ MP-202 can’t handle hold (inactive/sendonly) and codec changing in the same SIP INVITE
message.
„ A Reorder tone, indicating a registration error is played even if MP-202 doesn’t use a registrar (i.e.,
SIP proxy).
„ PRACK Disable does not function.
„ SIP messages from additional entities are blocked if the DST feature is enabled.
„ When “Connect on 180” is disabled and the MP-202 receives a 180 with SDP, the MP-202 doesn't
play a local dial tone.

Release Notes 16 Document #: LTRT-50511


Release Notes 4. Previous Version 2.6.1

4.3 Known Limitations in Version 2.6.1


Version 2.6.1 includes the following known limitations:
„ Faxes can be sent between the two local lines only if the selected fax transport mode is
Transparent (in G.711).
„ The Web interface is not automatically refreshed during the firmware upgrade process.
„ After the dial tone timeout has expired (and a fast-busy tone is played), the user can still make an
outgoing call.
„ A silence period of about three seconds is created 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.
„ When pressing ‘Flash’ + ’1’ or ‘Flash’ + ’2’ (for call hold and transfer), the DTMF is sometimes
heard at the remote side.
„ 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 a 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” message.
„ Enabling or disabling the Periodic Checking of Configuration File feature requires a reboot.
„ SIP NOTIFY with a “check-sync” event with a message body causes system restart even if there
are ongoing conversation calls.

Version 2.6.3 17 November 2008


MP-20x

Reader’s Notes

Release Notes 18 Document #: LTRT-50511


Release Notes 5. Previous Version 2.6.0

5 Previous Version 2.6.0

5.1 What’s New in Version 2.6.0


The following new features are supported in Version 2.6.0:
„ The MP-202 now supports SIP over TLS. TLS is a security mechanism that operates on top of TCP
enabling SIP entities to send and receive data in a secure and authenticated manner.
„ The MP-202 now supports STUN, a light-weight protocol that enables clients behind NATs
(Network Address Translators) to discover the presence and types of NATs and firewalls between
them and the public Internet. It also enables the client to determine the public IP addresses
allocated to the NAT. (For more information, see RFC 3489 STUN - Simple Traversal of UDP
through NATs.)
„ Mass Provisioning:
• The MP-202 enables downloading of an alternative format of the configuration file in “.ini-file”
format from a remote URL. When the MP-202 has a configuration file URL with the .ini
extension, it will expect it to have a flat .ini file-like format, for example:
rg_conf/voip/signalling/sip/proxy_address=10.16.2.4
rg_conf/voip/signalling/sip/proxy_timeout=3600
• The MP_202 enables displaying and saving of the current configuration in ini-file format: A new
checkbox was added to the configuration file Web page – “Display configuration in flat .ini-file
format”. When it is checked, the configuration file is displayed in flat .ini-file format. If the user
then selects to save the configuration file, it will be saved as a .ini file. The option '-i' in
rg_conf_print enables this at the CLI level.
• The MP-202 enables displaying and saving of only the modified configuration parameters: A
new checkbox was added to the configuration file Web page - "Display modified configuration
fields only". When it is checked, only the configuration parameters that have values other than
the default values are displayed and saved. The option '-m' in rg_conf_print enables this at
the CLI level.
• A new option was added to enable periodic checking for a new configuration file at the user-
configured URL.
„ A new method for establishing 3 way conference calls via a remote Conference server was added,
based on RFC 4240.
„ Improved Syslog debugging messages for diagnostic purposes. The debugging traces’ filter level
can be opened, closed or modified during run time.
„ SIP Security – the MP-202’s firewall can be configured to block incoming packets that have the SIP
signaling port as their destination. The user can configure up to two SIP entities (for example, the
SIP proxy or an SBC) which are not to be blocked by the firewall.
„ Addition of a Keep Alive option using SIP OPTIONS Messages. A SIP OPTIONS message is sent
periodically to the SIP registrar entity. The periodic interval can be configured from the
VoIP>Signaling Protocol Web-page.
„ Addition of a distinctive Call Waiting tone. When an INVITE message is received during a call, the
MP-202 applies a special call waiting tone based on the content of Alert-info header if it exists,
instead of the default tone. The alert data in an INVITE message defines the called party ringing
tone/call waiting tone.

Version 2.6.3 19 November 2008


MP-20x

„ 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:

5.2 Resolved Constraints in Version 2.6.0


The following bugs were fixed in Version 2.6.0:
„ CED detection is not received starting from the second T.38 session (DSP bug).
„ When dialing # & * busy tone was heard.
„ Enable sending # as a dialed digit.
„ The ptime attribute was removed from DTMF (101) in SDP body.
„ Enable CNG Detection: In Voice over IP Æ Voice and Fax the “Enable CNG Detection” check-box
was not operable. When the local fax machine connected to the MP-202 receives a fax, the MP-
202 switches to T.38 fax relay upon detection of the CED signal from the remote fax. If the local fax
machine sends a fax, the MP-202 switches to T.38 only after detecting the CNG signal from the
local side and the CED signal from the remote side. If the “Enable CNG Detection” checkbox is
enabled, the MP-202 switches to T.38 relay immediately upon detection of the CNG signal from the
local side, without waiting for the CED signal from the remote side.

5.3 Known Limitations in Version 2.6.0


Following are the known limitations:
„ Faxes can be sent between the two local lines only if the chosen fax transport mode is Transparent
(in G.711).
„ The Web interface is not automatically refreshed during the firmware upgrade process.
„ After the dial tone timeout has expired (and a fast-busy tone is played), the user can still make an
outgoing call.
„ A silence period of about three seconds is created 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.
„ When pressing ‘Flash’ + ’1’ or ‘Flash’ + ’2’ (for call hold and transfer), the DTMF is sometimes
heard at the remote side.
„ 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 a 2833 DTMF tone.

Release Notes 20 Document #: LTRT-50511


Release Notes 5. Previous Version 2.6.0

„ 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.

Version 2.6.3 21 November 2008


MP-20x Telephone Adapter
Release Notes
Version 2.6.3

www.audiocodes.com

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