RELEASE NOTE ETERNITY ME\GE\PE Card VoIP V2R9

Transcription

1 RELEASE NOTE ETERNITY ME\GE\PE Card VoIP V2R9 DATE: 24 th August, 2017 VERSION: V2R9 AUTHOR: Tapan Upmanyu

2 Contents Affected Configuration... 3 Release Information... 4 Up-gradation Procedure... 4 Packages... 4 Issues Fixed in this release... 5 Limitations... 9

3 Affected Configuration Will this firmware up gradation affects system configuration? NA Defaulted Parameters Sr. GUI Page Link Parameters Remarks No NA -- Removed Parameters Sr. No. GUI Page Link Parameters Remarks --- NA -- Relocated Parameters Sr. No. GUI Old Page Link GUI New Page Link Parameters --- NA --

4 Release Information Product Name Eternity GE/PE/ME/LE (VoIP Server Card) Product Variants Eternity GE/PE/ME/LE Version Revision V2R9 Date July 25, 2017 Up-gradation Procedure Step-1: Unzip ME_GE_PE_LE_VoIP_V2R9.1.0.zip file. Step-2: Delete all the contents of appl folder on VoIP Card FTP. Step-3: Copy all the Extracted contents to appl folder on VoIP Card FTP. Step-4: Restart the VoIP Card from Web Jeeves. Packages Field Package ETERNITY LE/ME/GE/PE File Name ME_GE_PE_LE_VoIP_V2R9.1.0.zip File Size

5 Issues Fixed in this release Conference is not working from Cisco Phone Eternity ME and Cisco CUCM is connected Peer to Peer. Cisco phones are registered with Cisco from Cisco phone. User dials to Eternity ME SLT user and press HOLD button from Cisco phone and proceeds to dial other numbers of Eternity ME User. After having speech, Cisco user presses the conference button but its not working. Only the original caller remains connected while others get disconnected. No Contact header in 2xx" error while registering SIP trunk SIP client sends Register request to the Server. Server responds with 200 OK + Contact header. As per general RFC concept, if Contact header is not received in the request, the client will not be registered. As per requirement, this issue is resolved by giving configurable option, which allows client to choose, which way to get registered (either With Contact header or Without Contact header). The option is available in Jeeves --> Under Configuration -> VoIP configuration -> SIP Trunk Parameters -> Allow registration for any contact. When this option is enabled the registration is done discarding the contact header and when disabled contact header becomes must with 200 OK. T38 UDPTL Fax Not working when Eternity VoIP server & SARVAM UCS are tied up. Eternity was tied up with SARVAM UCS over SIP Trunk. When Fax request for T38 UDPTL was initiated from Eternity side in this Eternity - SARVAM UCS tie up, Fax was not working on Eternity side.

6 Sometimes Sip Trunk status not updated properly when OPTIONS request is retransmitting and User disables the sip trunk When Sip Trunk gets disabled, VoIP Server will apply configuration in 10 sec. When Heartbeat is enabled and SIP trunk gets disabled, the status displayed is Heart Beat Failed instead of SIP Trunk disable. For Calls received on Sip Trunk, System uses CLI from Private Headers even if Anonymous CLI is not received in From Header. If the FROM header contains Non Anonymous information, the server does not have to check for the flag related to the Private header and display the CLI which is mentioned in the FROM header. But it is displaying the CLI which is mentioned in the Private Header. Hence, In an incoming SIP trunk Call, CLI was displayed wrongly as it was extracting the CLI from the Private Header instead of FROM header. When system gets CLI from Private headers for Any call, It will use same Private Header CLI for consecutive call. Make incoming call on SIP Trunk with anonymous CLI in From Header and Valid CLI in Private header. When Anonymous CLI is mentioned in FROM Header, the server needs to check for the flag related to the Private header. When enabled, the system gets CLI from Private header and when disabled it should get the CLI from the FROM header. This change of status of flag is not updated and hence the same Private header CLI is used for all consecutive calls till the system is reset.

7 In IPV4 and IPV6 switching case on LAN is not working. Sometimes Sip User gets registered with IPV6 using WAN interface instead of LAN even if WAN interface is not UP. SIP User VP110 gets registered with system as system WAN User even if it gets registered from LAN IPV6 network. In this case, VP510 call is not initiated and error message is displayed like Link down. For RTP Mode Trans-coding, VoIP server Application gets crashed when INVITE request without SDP Offer and with 100rel received. When an OG call from SIP Extension is done, It generates INVITE Request without SDP offer and with 100rel. Now, when called SIP Extension answers the call, system should generate 200OK Response to INVITE Request with SDP offer. While generating SDP offer VoIP server application gets crashed. In user to user call, by doing multiple times hold unhold sequence, call is disconnected. A call between 2 extensions is established. Now, from any one extension if HOLD/UNHOLD sequence is done multiple times, Call is disconnected sometimes. VoIP Server is not sending G.722 codec in outgoing call invite request. VoIP Server is not sending G.722 codec in outgoing call invite request even if it was set in the hardware template.

8 In Delay offer on trunk, transcoding mode - hold the call then unhold it, system generates ACK without sdp,so remote system disconnect the call. System Does not send SDP Answer in ACK request for Multi Coder Re-INVITE when Delayed Offer flag is enabled on Sip Trunk When UNSUBSCRIBE request for MWI received, System is not sending NOTIFY (terminate) to UAC. When User Agent Client gives UNSUBSCRIBE request for Message Wait Indication to the system, it is not notified to the client every time and hence it does not unsubscribe the UAC. Refresh and unsubscribe of presence subscription is not proper handled. In Presence Subscription when total system watchers reached to 3 and new watcher is added, When new added watcher send refresh or Unsubscribe request then it is rejected by 403 forbidden When TCP Enable flag kept disabled, VoIP server application gets crashed on power on. At power on the VoIP server application gets crashed when TCP Enabled Flag is kept disabled.

9 Limitations Authentication for Sip user - Authentication Id must be unique for the system (Sip Extension and Digest Authentication table). In case of duplication/conflict, system admin has to resolve Authentication Id conflict by changing all similar Auth IDs of Sip Extension/Digest Authentication parameter. Video Call - Use H264 codec for Video calls. Quality of Video might get degrade if all Video calls are with H263/H263+ codec. RTP Relay/DRTP - VP248 open sip client is not supported in RRTP/DRTP OPTIONS - VoIP Server does not support 200 OK with SDP against received OPTION (SIP Server stack limitation). SDP - Domain in c-line of SDP is not supported SIP - System ignores any sip message having packet size more than 2048 bytes. Remarks: Due to memory concern, It will not be resolved in Eternity variants. Audio Codes DSP - a. Audio Codes DSP module does not support ptime (Packet Time) above 40ms. It will use ptime as 20ms for that. b. T38 RTP fax is not working between audio codes version 640 (VoIP Server Version V1R7 for Eternity GE/ME/PE) to 660 (VoIP Server Version V1R8 onwards for Eternity GE/ME/PE) & vice versa, but working fine between audio codes version 660 (VoIP Server Version V1R8 onwards for Eternity GE/ME/PE). c. Fax in Pass through mode will not work properly if dynamic payload is used as FAX payload (PCMA/PCMU). d. Comfort Noise will not be generated properly in case of dynamic payload is used for CN (Comfort Noise).