This document describes new features, the latest hardware and software requirements, known limitations, and other pertinent release information for the latest release of EdgeMarc VOS 16.4.0.
The EdgeMarc VOS 16.4.0 documentation is located at the following Ribbon Wiki space: EdgeMarc VOS 16.4.x Documentation Home.
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The EdgeMarc is a flexible, easy-to-use session border controller that provides critical networking functions for IP-based voice and data. EdgeMarc VoIP Operating System (VOS) provides award-winning features for converged networking environments. These features enable the EdgeMarc to act as an enterprise session border controller and demarcation point for managed services.
|New Features, Improvements, Issues Resolved|
EdgeMarc VOS version 16.4.0 supports the following platforms:
Refer to EdgeView and EdgeMarc Interoperability Matrix for a detailed EdgeMarc VOS interoperability matrix.
The following new features are included in this release:
Privacy header handling
A GUI option has been included such that the user can configure whether the SIP server(s) is trusted or not. EdgeMarc should pass through a privacy header in inbound INVITE to outbound INVITE in case the outbound SIP server or trunking device is trusted. EdgeMarc should remove a privacy header from INVITE before forwarding it to the outbound trunking device in case that outbound SIP server or trunking device is NOT trusted.
For detailed information, refer Best Practice - Configuring B2BUA with Header Manipulation Rules.
MS Team - SIP-ALL FQDN - Change
The Edgemarc supports updates to the trusted MS Teams server lists per Microsoft request.
For detailed information, refer the section Configure the SBC for Microsoft Teams Direct Routing#ConfiguringtheB2BUAandHeaderManipulationrules.
SIP OPTION Pass through to PBX
The Edgemarc can now respond to SIP Options requests received from the IP network based on the status of a configured PRI.
For detailed information, refer ISDN PRI and Survivability.
Add VRRP support to the EM-7300
The Edgemarc 7300 series now supports VRRP.
Receiving ARP Trap
The Edgemarc can now communicate the LAN side ARP table contents to an Edgeview that is running code with corresponding support. This feature requires Edgeview 16.4.
Message Translation feature for 180/183 with/without SDP
The Edgemarc GUI contains new options to control SIP message translation between 180 and 183 messages.
For detailed information, refer Configuring Message Translation of 180/183 with or without SDP, and Use VoIP ALG settings.
SIP Proxy Load Balancing (Round Robin)
The Edgemarc now supports outbound load balancing across SIP servers.
Enable sysloging of PRI from EdgeView
When the Edgemarc is used with a compatible Edgeview, ISDN logging can be enabled and disabled via Edgeview. This feature requires Edgeview 16.4.
The following software versions are required for this release and are available for download from the Customer Portal:
|EM VOS 16.4.0|
Use the EdgeMarc WebUI to verify the currently installed software version.
This section contains specific upgrade notes for customers planning to upgrade to EdgeMarc VOS 16.4.0.
|Issue ID||Case No.||Sev||Problem Description||Resolution|
|220119-408745||1||Three way conference calls with one or more SRTP legs are not processed correctly.|
SRTP calls now correctly decrypted in a conference environment.
|211115-400034||1||Phones that are registering without 'Expires' header were unable to register after the initial Register. The root cause is that the re-Registers without expires header were treated as de-register in EM.|
If the received register doesn't contain an expires header or expires parameter in contact header, it will be treated as a 'register' instead of 'de-register'.
|211111-399522||1||Multiple instances of codecs in SDP are not properly stripped with 'SDP Modification' configuration. Internal logic in EM was checking only for a single occurrence of a codec while stripping.|
Fix is done to handle 'codec-delete' to properly strip the codecs with multiple instances in SDP.
The Edgemarc was not routing SIP REGISTER messages properly in some survivability failover scenarios.
Corrected handing of REGISTER messages in failover cases.
|EM-27049||220304-562470||1||The "401 Unauthorized" messages for SUBSCRIBE were not getting forwarded to the client. Also while monitoring mandlogs, mutex logs were observed immediately after receiving SUBSCRIBE.|
The below fixes were done -
Race condition/Mand crash when the challenged REFER request got 403 response.
Correction is done by assigning right transaction in the statemachine.
|220112-407804||1||EM was not sending REGISTER to SBC with a new call-id even after the client sends a REGISTER with a new call-id.|
Fix done to reset the register core, if the call-id changes for a client.
|220411-572785||1||No Ring Back Tone provided for outbound call from PRI when 183 received without SDP.|
Fixed logic to send alert while receiving 183 without SDP as well, similar to 180 without SDP.
EM2009a refer domain used an not proxy ip.
Corrected an issue where the Edgemarc used an incorrect IP when translating SIP Refer messages.
|Issue ID||Case No.||Sev||Problem Description||Resolution|
|220311-564416||2||Incorrect value for priority in the SIP Server Reachability table in the Survivability page. The root cause is that the priority value was stored in a variable that could not accommodate its 0-65525 range. Its variable type was unsigned char, which can only accommodate a 0-255 range. Thus, if the priority was over 255, the value could not be stored correctly.|
Increased the size of priority in DNS structures to accommodate its 0-65535 range. A similar change was made for weight because it was also too small for its same range.
|EM-27173||220413-573629||2||Calls intermittently have incorrectly mangled Request URI line.|
Change is done for EM to not preserve the Contact header from OPTIONS response.
|EM-27366||220622-592349||2||EdgeMarc incorrectly translating SIP REGISTER messages. The EdgeMarc continues to use the same address from the first REGISTER message it received for a particular DID regardless of external SIP device config changes including either change in request domain or call-id from LAN.|
EdgeMarc will recreate a new SIP REGISTER cores with updated target and domain information for each new REGISTER anytime when there is a change in the request domain or SIP Call-ID.
|3||Not able to log in to EM GUI when browsing via an FQDN when session management is enabled. The root cause is that the domain field could not pass when enabled session management.|
Ignore the domain field.
Workaround: An upgrade is required to resolve this issue.
|3||If Remote and Local URI Domain headers contain "anonymous.invalid" as domain, the EdgeMarc will use the custom domain from SIP Settings Page in REFER-TO header and local uri in REFERRED-By header respectively.|
Made the fix by adding sipserver domain in REFER-TO header and local uri in REFER-BY header if the remote uri is "anonymous.invalid"
RTP streams with invalid time stamps were observed on EM2900 running TCP SIP and HA. An improper value set to the protocol field while replicating RTP stream information to the standby system caused this issue.
Protocol information, RTP entry start, and last received timestamp information will be updated on standby every 60 seconds. Also updated debug logs to print the last RTP received time.
|EM-26962||Security - OpenSSL - New Version - Port - 1.1.1m||2||OpenSSL has been updated.|
Code is fixed to updated the OpenSSL.
There are no known issues of high severity in this release.
|Issue ID||Case No||Severity||Problem Description||Workaround|
|EM-27420||-||2||TLS registration of an FXS port on an ATA 302/304 does not work when IPv6 SIP proxy is specified. TLS with an IPv4 SIP proxy is functional as is UDP and TCP transport for IPv6 SIP proxies.||N/A.|
|EM-27418||-||2||After an upgrade from VOS version 15.8, configuration of VoIP and SIP parameters may become blocked if the SIP Server Transport is configured as Pass Through and one or more SIP transports is set to blocked on the VoIP page. A warning on the VoIP and GUI pages is displayed. As a workaround, change the SIP Server Transport setting to match the SIP transport protocol in use.||N/A.|
|EM-27451||-||2||In a LAN (RTP) to WAN (SRTP) call, multiple changes in codec in the media/audio stream during early media phase (before the call connects) can result in one way audio once the call connects.|
Switch to RTP. This problem is not observed in RTP to RTP calls where the media is not encrypted.
|EM-27416||-||3||A call destined for an FXS port that is processed by the B2BUA function does not ring with the ring pattern selected for the port on the SIPUA → Advanced page. The port uses only the default ring pattern.||N/A.|
|EM-26273||-||3||An incorrect Lost Packet counter may be displayed on the Survivability page if the configured SIP server can not be resolved by DNS.||N/A.|
|EM-26923||-||3||The EM accepts the following FQDN: some_fqdn.com but does not resolve it. Nor does an error indicate the Underscore character is illegal character according to RFC 1123.||N/A.|
|EM-27435||-||3||Inbound call from Microsoft Teams to SIP phone and then subsequently transferred (blind or consultative) to another SIP phone can lead to one-way audio. This occurs with media bypass disabled. It is not observed with media bypass enabled.||N/A.|