Page 1 Voice Over IP Manual...
Page 2: Table Of Contents
Table of Contents Part I Part 1: VoIP Reference Manual 1 1. What is Voice over IP? (VoIP) ........................... 4 2 2. Factors Affecting Voice Quality ........................... 6 3 3. Implementing QoS ........................... 11 4 4. Other Issues Affecting VoIP ...........................
Page 3 Contents Testing the SV9100 Netw ork Connection ................................. 254 8 Appendix A: TCP_UDP Port Numbers ........................... 255 9 Appendix B: SIP Configuration Example ........................... 257 10 Appendix C: ToS Field Values ........................... 260 11 Appendix D: Configuration of External DHCP Server ...........................
Page 4: Part I Part 1: Voip Reference Manual
To allow this interoperability each system must adhere to the same standard and “talk the same language” (known as a protocol). There are various standards and protocols that are used. The most common types (and those used by NEC equipment) are:...
Page 5 Part 1: VoIP Reference Manual Session Initiation Protocol (SIP) SIP is a proposed standard, developed by the Internet Engineering Task Force (IETF) for setting up sessions between one or more clients. It is currently the leading signalling protocol for Voice over IP, gradually replacing H.323 in this role.
Page 6: 2. Factors Affecting Voice Quality
Note: Not all CODEC’s listed above are available for all applications. Compatibility? It should be noted that NEC does not guarantee that any third-party equipment will operate correctly with NEC equipment. 2. Factors Affecting Voice Quality Quality of Service (QoS)? Quality of Service (QoS) is one of the most important factors for VoIP.
Page 7 Part 1: VoIP Reference Manual The buffering, queuing, and switching or routing delay of IP routers primarily determines IP network delay. Specifically, IP network delay is comprised of the following: Packet Capture Delay Packet capture delay is the time required to receive the entire packet before processing and forwarding it through the router.
Page 8 cause “choppy” speech as the WAN link will suffer from congestion. In this case, a lower bandwidth CODEC (such as G.729) may be more appropriate. Bandwidth? Available bandwidth has a major influence on voice quality in VoIP networks. Bandwidth is usually expressed in the number of bits per second (bps) that can be transmitted over a network link.
Page 9 Part 1: VoIP Reference Manual Example two: CODEC: G.729 Frame Size: 80ms Voice Frames per Packet: 8 Voice Sample Size: 80ms (frame size x Voice Frames) Bandwidth Required: 12Kbps Note: Figures above do not take into account Layer 2 overhead. This varies dependant on the Layer 2 protocol used (e.g.
Page 10 Layer 2 Protocol? All VoIP packets are encapsulated into a Layer 2 protocol (e.g. Ethernet, PPP, Frame Relay) for transmission over the data network. Every Layer 2 protocol has different characteristics and header sizes. This means that the amount of bandwidth will vary dependant on the protocol used. bandwidth graph above shows the affect of different Layer 2 protocols on the bandwidth requirement.
Page 11: 3. Implementing Qos
Part 1: VoIP Reference Manual SDSL 1Mbps – 10Mbps - Symmetric bandwidth - Availability – only certain areas are (Symmetric Digital Subscriber Line) SDSL enabled - Contention (shared bandwidth) - No bandwidth guarantees - Delay varies dependant on Internet conditions - Unreliable BT Baseband Depends on cable...
Page 12 In this case there is only one host on each end of the network which is unrealistic. In reality there would be many hosts all sending data over the narrow bandwidth. This means that the routers must buffer the packets and transmit them over the Kilostream as efficiently as possible. When this occurs, certain packets will be dropped by the router and some packets will be delayed.
Page 13 Part 1: VoIP Reference Manual Once the router has been configured for QoS, it will examine incoming packets and allocate the relevant priority to the packet. The diagram below shows the effect that Priority Queuing would have on “voice” and “data” networks. The packets arrive randomly and are processed and output according to the QoS policy –...
Page 14 addresses, and is usually handled by Routers. However, sometimes it is necessary to implement Layer 2 QoS – usually in large LAN environments with many IP phones. Layer 2 devices work with Ethernet frames (encapsulated IP packets) rather than IP addresses. These devices are usually Switched Hubs (Switches).
Page 15 Part 1: VoIP Reference Manual VID - VLAN ID is the identification of the VLAN, which is basically used by the standard 802.1Q. It allows the identification of 4096 VLANs. Length/Type - This field indicates either the number of MAC-client data bytes that are contained in the data field of the frame, or the frame type ID if the frame is assembled using an optional format .
Page 16 Layer 3 QoS QoS is most commonly implemented at Layer 3. This allows the VoIP packets to be prioritised by routers, before they are forwarded to their next hop. Layer 3 QoS uses the Type of Service (ToS) field of the IP packet. This is an 8 bit field in the header of the IP packet.
Page 17 Part 1: VoIP Reference Manual Marking is the first step in this process and is often the only step that the NEC dealer will perform. Protocol Structure - IP/IPv4 Header (Internet Protocol version 4) Version - the version of IP currently used.
Page 18 complete. Header Checksum - Helps ensure IP header integrity. Since some header fields change, e.g., Time To Live, this is recomputed and verified at each point that the Internet header is processed. Source Address - Specifies the sending node. Destination Address - Specifies the receiving node. Options - Allows IP to support various options, such as security.
Page 19 Part 1: VoIP Reference Manual Normal throughput High throughput Reliability Value Description Normal throughput High throughput Diffserve (Differentiated Services) Differentiated Services (Diffserv) is a method of utilising ToS field in an IP header. Diffserv is now commonly used instead of IP Precedence as it provides greater flexibility. This method uses 6 bits of the ToS field to determine the priority –...
Page 20 Comparison of IP Precedence and Diffserv Values As stated earlier, IP Precedence and Diffserv use the same field in the IP header to mark packets. It is possible to have the same ToS value for either method which means that the two methods can work alongside each other.
Page 21 Part 1: VoIP Reference Manual 000011 000100 000101 000110 000111 001000 Class Selector 1 001001 001010 AF11 (Assured Forwarding) 001011 001100 AF12 (Assured Forwarding) 001101 001110 AF13 (Assured Forwarding) 001111 010000 Class Selector 2 010001 010010 AF21 (Assured Forwarding) 010011 010100 AF22 (Assured Forwarding) 010101...
Page 22 However, this sample configuration has been provided as it is a common scenario and is a good example of how QoS can be implemented on a router. NEC do not endorse or provide support on any third party equipment unless it is supplied by NEC.
Page 23 Part 1: VoIP Reference Manual The configuration file below shows the London Cisco 2621 router configuration. Some unrelated configuration has been removed. A description of some of the key commands can be found below the configuration. Cur r ent c onf i gur at i on : 2023 by t es v er s i on 12.
Page 24: 4. Other Issues Affecting Voip
3) Defines a Policy Map called VoIPPolicy 4) Creates a Class called VoIPClass and assigns this to the VoIPPolicy 5) Allocates 50Kbps of bandwidth to the VoIPClass 6) and 7) Determines that any data that does not match VoIPClass should be processed using the “fair-queue”...
Page 25 Summary: When implementing VoIP using internet based connections it is very important that these factors are considered, and that the customer is made aware that you (the installer) or NEC cannot be held responsible for any quality issues experienced. Firewalls Network security is always a concern when connecting the LAN (Local Area Network) to the WAN (Wide Area Network).
Page 26 This re-routing of IP address from one address (Private Address) to another (Public Address) and the allowing of only selected ports to be opened create problems for the VoIP. When a VoIP terminal receives a VoIP packet from a far-end site, the voice application routes information back to the far-end based on the embedded address.
Page 27 Part 1: VoIP Reference Manual show the tunnels that have been created through the internet. Each network can connect to the others as though they are connected with private connections (kilostream, etc.), without the issues relating to NAT. Conclusion When IP address translation is applied to a VoIP packet, the application fails and the communication path is broken.
Page 28: 5. Troubleshooting
It is highly recommended that any VPN hardware used for VoIP has the facility to prioritise voice traffic. 5. Troubleshooting This section aims to provide some helpful tips to perform Voice Over IP troubleshooting. The first step in resolving any issues would be to read through the relevant manuals/documentation and refer to any example configurations.
Page 29 Part 1: VoIP Reference Manual Unsuccessfull PING If you are unable to ping a device it may mean that either the source or destination device: is not configured correctly is not connected to the LAN (e.g. cable disconnected) has a developed a fault or any device in between the source or destination may be faulty (e.g.
Page 30 Note that there are other options available with the Microsoft Windows implementation of ping. The most commonly used options are listed above. Examples: ping 192.168.2.100 -t Continually pings 192.168.2.100 until Ctrl-c pressed ping 192.168.2.100 -n 10 -l 40 Sends ten 40-byte packets to 192.168.2.100 ping 192.168.2.100 Sends four 32-byte packets (default) to 192.168.2.100 Pinging from an DT700 or DT800 IP Phone...
Page 31 Part 1: VoIP Reference Manual that point fails. This would suggest that the Leased Line or remote router has a problem. The local PC (192.168.1.101) can ping all devices apart from the DT700. DT700 cannot ping anywhere. This would suggest that there is a problem with the DT700 or its connection to the switch/hub.
Page 32: 6. Glossary
6. Glossary 802.1p An IEEE standard for providing QoS using three bits (defined in 802.1q) to allow switches to reorder packets based on priority level. 802.1q An IEEE standard for providing virtual LAN (VLAN) identification and QoS levels. Three bits are used to allow eight priority levels, and 12 bits are used to identify up to 4,096 VLANs.
Page 33 Part 1: VoIP Reference Manual Compression Compression is used at anywhere from 1:1 to 12:1 ratios in VOIP applications to consume less bandwidth and leave more for data or other voice/fax communications. The voice quality may decrease with increased compression ratios. Connection-oriented Mode of communication in which a connection must be established between the transmitter and receiver before transmission of user data.
Page 34 in its G-series recommendations. G.723.1 Describes a compression technique that can be used for compressing speech or audio signal components at a very low bit rate as part of the H.324 family of standards. This CODEC has two bit rates associated with it: 5.3 and 6.3 kbps. The higher bit rate is based on ML-MLQ technology and provides a somewhat higher quality of sound.
Page 35 Part 1: VoIP Reference Manual Pulse Coded modulation. Quality of Service. Measure of performance for a transmission system that reflects its transmission quality and service availability. Standards based QOS for VoIP usually involves the implementation of Ethernet standards 802.1p and 802.1q at layer 2 across an Ethernet. At layer 3, the IP standard Diffserv defines bits settings in the TOS (type of service) in the IP header which will identify packets as being associated with a specific service.
Page 36: Part Ii Part 2: Voip Manual
The UNIVERGE SV9100 system uses IP for various applications. This section describes the procedure for connecting the UNIVERGE SV9100 system to an existing data network and configuring TCP/IP. This is the first step in implementing VoIP and other IP applications.
Page 37 DHCP as there is no need to assign and program individual IP addresses for the LAN equipment. To use a dynamic IP address, a DHCP server must be provided. The UNIVERGE SV9100 system GCD-CP10 blade provides an internal DCHP server, enabling the ability to use DHCP.
Page 38 Assume that a UNIVERGE SV9100 is added to the existing data network. The Network Administrator (or IT Manager) should provide the following: IP Address (for the GCD-CP10 blade) IP Addresses (for the GPZ-IPLE daughter board) Subnet Mask Default Gateway A spare switch/hub port First, program the UNIVERGE SV9100: 192.168.1.200...
Page 39 Windows Server) or static addressing (as illustrated in Example Configuration 1). However, if the UNIVERGE SV9100 is to be installed on a new network the Network Administrator may want to use the UNIVERGE SV9100 internal server (this is called inDHCP). In this example, the client PCs get an IP address, subnet mask, and default gateway from the inDHCP server.
Page 40 Now connect the UNIVERGE SV9100 GPZ-IPLE Ethernet Port to the switch/hub port, using a standard CAT- 5 patch cable. The UNIVERGE SV9100 is now configured on the network and its DHCP server is ready to allocate IP addresses. The client PCs should be set to Ob ta in IP Ad d re ss Au to m a tica lly .
Page 41 Part 2: VoIP Manual terminal used. If the system is set to allow peer-to-peer calls between standard SIP stations then no system DSP resources are required to support the video functionality between standard SIP phones. Conditions CO calls cannot be transferred to a Remote Conference pilot and must directed to the conference pilot as a DID or DIL termination.
Page 42 daughter board must be reset for the changes to take effect. This will happen automatically once all VoIP resources go idle. Up to 256 VoIP DSP resources are available when the SV9100 is properly licensed. Video streaming is not supported to SIP Terminal via Netlink. Simple MCU video is not supported for SIP Trunk calls.
Page 43 Part 2: VoIP Manual IPLE Video Streaming Termination Video functionality can be configured to use system VoIP DSP resources for non-MCU video calling if peer-to-peer calls between standard SIP phones is either disabled or video codecs not supported between the terminals being used. If the system is set to allow peer-to-peer calls between standard SIP stations then no system DSP resources are required to support the video functionality between standard SIP phones.
Page 44 Licenses Conditions The system will always reserve 64 of the total 256 VoIP DSP resources for voice calls. If peer-to-peer is disabled the standard SIP video feature requires VoIP DSP resource be reserved for this function reducing the number of VoIP resources available for SIP phone calls. When set in programming the following tables show how many resources are reserved for each video mode.
Page 45 Part 2: VoIP Manual VoIP LAN Link Speed Information (Release 3) The SV9100 can now display the LAN link information for duplex and speed of either the CPU or VoIPDB interfaces. Easy Edit – Advanced Items/VoIP/General Settings/VoIP Configuration/VoIP LAN Link Speed Information LAN Link Speed of CPU - Displays the duplex and speed of the LAN connection between the CPU interface and network switch.
Page 46 Store Statistical Information of RTP (Release 3) The SV9100 can now store statistical information of RTP on the SD Card. This information once stored on the SD Card can then either be retrieved using PC Pro to download or by saving to a USB stick using telephone programming.
Page 47 Part 2: VoIP Manual Limitations When the remote device does not support RTCP protocol, the statistical information of the RTP communication of the other party cannot be obtained. In Peer to Peer, the statistical information of RTP communication is not obtainable. When a write error occurs during the writing to file on the SD Card, the RTP statistical information data under writing is canceled.
Page 48 Ext/TRK Number The Extension or Trunk Number on the system side. Ext/TRK Number(Call partner) The Extension or Trunk Number on the remote side. Cumulative Lost Cnt Receiving RTP packet loss total The Max rate of packet loss When the max packet loss rate and the average packet loss Max Fraction Lost rate are the same value, we can think a packet loss occurs constantly.
Page 49 Part 2: VoIP Manual The max Interval of the arrival time of a RTP packet. Max Inter Arrival Jitter Fluctuation between the arrival time of a RTP packet. Since it is dependent on a sampling frequency, a unit is set to ms by dividing by 8 in the case of G.711 (8-kHz sampling).
Page 50 Operation Procedure to save Statistical Information of RTP to SD Card by Service Code. Procedure of extraction Statistical Information of RTP (TEL Pro)
Page 51 Part 2: VoIP Manual Procedure of extraction Statistical Information of RTP (PC Pro) From the Tools menu click RTP Information.
Page 52 Specify the location the file(s) will be downloaded to and press OK.
Page 53 Part 2: VoIP Manual Click OK Saving will be completed and the files stored at the location selected.
Page 54: Qos Settings
QoS standards: IP Precedence and Differentiated Services (Diffserv). Although IP Precedence and Diffserv are both supported on the UNIVERGE SV9100, it is becoming more common to use Diffserv only. The two methods of QoS are interoperable (for example IP precedence values can be mapped to Diffserv values and vice versa).
Page 55 Part 2: VoIP Manual Diffserv – Use only if the ToS Mode is set to Diffserv. Enter the required Diffserv value between 0 and 63. Below is a table showing which QoS item is required for each particular VoIP feature of the SV9100. AspireNet Standard H.323...
Page 56: Ip Address Collision
Be aware that the RTP/RTCP Layer 3 QoS setting is common to all signalling protocols so if there are several VoIP protocols being used on the same system, the speech cannot have different QoS values per protocol. i.e. If an SV9100 is programmed to use NetLink and SIP trunks, the speech will use the same RTP/RTCP QoS value whether a NetLink call is in progress or a SIP trunk call is in progress.
Page 57 Part 2: VoIP Manual...
Page 58: Iple Blade
Conditions/Comments The collision alarm will continue until the IP address conflict is resolved. The collision detection will only occur in the same network subnet as the SV9100. The G-ARP packets are sent every 5 minutes, this timer is not programmable. If multiple IP address collisions are detected only one will be shown on the Alarm Display Telephone.
Page 59 Part 2: VoIP Manual There are several different codec’s available for use on the SV9100 as listed below. G.711 (64kbps) G.722 (64kbps) G.726 (32kbps) G.729 (8kbps) Not all of these codec’s are available for all VoIP applications, refer to the relevant section of this manual for further information.
Page 60: Iple Installation
Connect the IPLE daughter board to the CD-RTB or to an external switching hub using an Ethernet cable. Refer to the UNIVERGE SV9100 Programming Manual for detailed programming instructions. GPZ-IPLE Switch Settings This daughter board does not have any switches that need to be set and does not require any hardware...
Page 61: Iple Led Indications
Part 2: VoIP Manual 2.4.2 IPLE LED Indications GPZ-IPLE LED Indications LED indications for the GPZ-IPLE Daughter Board are indicated in Table 1 - IPLE Daughter Board LED Indications on page 4-29. Each LED is listed with its associated function and LED and Operational status. Refer to Figure 1 - GCD- CP10 Blade with Daughter Boards Installed on page 4-16 for the location of the LEDs on the blades.
Page 62: Iple Voip Channel Licensing
2.4.3 IPLE VoIP Channel Licensing The GPZ-IPE VoIPDB installed in a SV9100 system has a maximum channel (DSP) capacity of 256. A default system with no free license enabled or additional licenses installed has 8 VoIP channels available. This means 8 IP devices could make calls through the GPZ-IPLE to TDM extensions and trunks.
Page 63 Part 2: VoIP Manual The 8 free VoIP channels are available on this SV9100. The below table illustrates the number of VoIP channels that are allocated in licensing with each IP device. These would be visible in the Feature Activation screen when licenses are installed as VoIP Channels (5103).
Page 64 Select the GCD-CP10 slot for the standalone system. If Netlink is used then the Primary GCD-CP10 and Secondary GCD-CP10(s) will require VoIP channels allocated depending on the number of IP devices used at each node. This is flexible but most likely will match the number of IP devices used at each node.
Page 65 Part 2: VoIP Manual Based on the IP license table further up this means the SV9100 system will have the following VoIP channels allocated. Feature Feature Quantity VoIP Channels Code...
Page 66 0002 Netlink 5001 IP Trunk 5012 K-CCIS over IP 5091 Networking over IP (AspireNet) 5101 DT IP Terminal 5111 IP Terminal There are 4 less VoIP channels allocated for DT IP Terminals (5101) and IP Terminals (5111) because the quantity displayed in the Feature Activation screen for each of these also includes the 4 free terminal licenses, however the 8 free VoIP channels (5103) are not displayed on this screen.
Page 67 Part 2: VoIP Manual After the system has initialised the cards, the new VoIP channels should be allocated on the IPLE and can be confirmed by pressing Feature + 4 on a keyset.
Page 68: Extensions
SIP is an industry standard protocol and therefore there are many different hardware and software based phones. As these phones are not developed by NEC and are not designed specifically for use on the SV9100, they do not support majority of the features that you would find on an SV9100 Keytelephone.
Page 69: Sip Mlt
Optional LCD panels, keypads, handset cradles, face plates and coloured side panels can easily be snapped on and off. See the tables for compatible kits. SIP MLT extensions use an enhanced version of the SIP protocol (iSIP) developed by NEC to communicate with the SV9100 system.
Page 70 DT800 DT700 to register is set to 1 (default) Configure an SIP MLT and connect to the LAN The SIP MLT will take port 17 (ext 216 in default) Port 18 will also be reserved for use by another IP extension. If a second extension card (e.g.
Page 71 Part 2: VoIP Manual Press 2 2 1 - Enter the IP address of the SV9100 GPZ-IPLE card Press OK. Press Exit. Press 4 1 - Enter 5080 (SIP Server port) Press OK. Press Exit several times to get to the main menu. Press Save.
Page 72 Registration Procedure – Manual This mode of registration gives the ability to ‘Hot Desk’ from one SIP MLT to another. This means you can move from one handset to another and keep the same extension number and relevant programming. When the Register Mode is set as Manual in Easy Edit - Advanced Items/VoIP/Extensions/DT800 DT700 Setup./DT800 DT700 Hot Desk/DT800 DT700 Logon Type.
Page 73 Part 2: VoIP Manual The SIP MLT will now attempt to register to the SV9100. If the login information is accepted the display will change to normal idle status. If the extension you are trying to log on as is already in use you may be prompted with ‘Override?’ On the display with a ‘Yes’...
Page 74 network. Furthermore, if the original system recovers the handset may revert back to its original registrar (software release 4 feature). To set the registration destinations and port numbers for each SIP MLT follow these steps: - Press Menu 0 User Name - ADMIN Password - 6633222 2.
Page 75 Part 2: VoIP Manual Available Codecs for SIP MLT handsets · G.711 64Kbps codec MOS 4.4 · G.722 64Kbps codec MOS 4.4 · G.729 8Kbps codec MOS 4.0 The bandwidth values quoted for these codecs are for the digitized speed in one direction only. The actual bandwidth required for a call will depend on many other factors and will be much higher than these figures.
Page 76: Sip Mlt Models
The ‘QoS’ option gives information about lost packets, codec in use and the payload size. The ‘System Information’ option gives various information about the network settings, SIP Settings, Audio & Visual Settings, Maintenance Settings and Terminal Information. The ‘Ping’ option gives the ability to ping another IP address to check connectivity across the network. Note: The lost packet count can only be updated when the SIP MLT has connectivity with the SV9100, if connection is lost during the call the display will not be updated.
Page 77: Sip Mlt Features
Part 2: VoIP Manual ITL-12PA-1. This IP value Multiline Terminal with Analogue Power Failure adapter has 12 line keys with display. ITL-24D-1. This IP value Multiline Terminal has 24 line keys with display. ITL-32D-1. This IP value Multiline Terminal has 32 line keys (24 line keys plus an eight line key LK Unit) with display.
Page 78 Encryption - Handset Programming On each DT800/DT700 Enter configuration menu (menu 0) followed by the password. Option 2. SIP Settings. Option 7. Encryption. Option 1. Authentication Mode = Enable Option 2. One Time Password = (As entry in PRG10-46-09) - Save The handset will reboot and attempt to authenticate with the SV9100, if it is successful it will show 'Authentication Accepted'.
Page 79 Part 2: VoIP Manual (e.g. via the internet). This means a VPN is no longer required to place a DT800/DT700 on a remote network. (This feature is not available for 3 Party SIP devices (standard SIP)) A network configuration diagram is shown below Depending on their locations, terminals are classified into the following four categories: LAN terminals Terminals installed on the LAN where the main device is installed.
Page 80 Easy Edit – Advanced Items/VoIP/Extensions/DT800 DT700 Setup/DT800 DT700 NAT/DT800 DT700 NAT Exempt Networks. (PRG10-58) Enter the network address and subnet mask of DT800/DT700 LAN terminal via local router (if connected – see below for further information). Port forwarding system side On the router connected to the SV9100 network, forward port UDP/5080 and UDP/5081 to the SV9100 GPZ- IPLE IP address.
Page 81 Part 2: VoIP Manual router on the network where the handset belongs. Port forwarding handset side On the router connected to the handset network, forward port UDP/5060 to the IP address of the handset. Also forward port UDP/3462 to the IP address of the handset. If you have multiple NAT handsets on the same remote network you will need to change the port numbers on each handset so that they are unique on the network.
Page 82 In this example there would be the following port forwarding rules in the NAT router connected to the SV9100: - UDP/5080 -> 172.16.0.10 (DT800/DT700 signalling to SV9100) UDP/5081 -> 172.16.0.10 (DT800/DT700 signalling to SV9100) UDP/10020 – 10051 -> 172.16.0.20 (DT800/DT700 RTP/RTCP to Primary SV9100) UDP/20020 –...
Page 83 IP addresses for terminals have to be specified statically. When allocating an IP address using DHCP, the IP address might change. NEC does not guarantee proper operation in this case. If installing multiple terminals in the domain of the NAT router on the terminal side, the SIP port number and RTP/RTCP port number for each terminal must be specified so as to avoid overlapping.
Page 84 The SIP server cannot be switched. (Only one address can be registered as the SIP server.) SIP MLT NAPT Improvement for Registration Timers Sometimes when a DT800/DT700 Terminal connects to the SV9100 via NAPT the intermediate router may have a firewall function, if there is no communication for a defined period the router may close the port used by the DT800/DT700.
Page 85 Part 2: VoIP Manual 1~144 ~192 ~180 ~512 15-05-47 15-05-48 SIP MLT NAPT Enhancement The Dynamic NAPT (Network Address Port Translation) feature has been enhanced to improve the performance of SIP MLT when used with a dynamic router connected at the terminal side. Port forwarding on terminal side should no longer be necessary when using this feature with a suitable dynamic NAT enabled router.
Page 86 Exit and save, the handset will now try and register to the SV9100. If it fails it may be because the router does not support dynamic address and port mapping. If this is the case Static NAT may have to be used. SIP MLT Time Zone Offset If a SIP MLT is located in a different time zone to the system it is registered to the time and date may be incorrect.
Page 87 Part 2: VoIP Manual Set to Multicast, Unicast or Auto on a per extension basis. Related programming PRG10-19 GPZ-IPLE DSP Resource Selection. Three new options, 6 – Common without Unicast Paging, 7 – Multicast Paging, 8 – Unicast Paging. Limitations One DSP is required for each extension using Unicast Paging Mode where as only one DSP is required for all extensions using Multicast Paging Mode.
Page 88 Example: Chassis id: 172.16.0.100 (Terminal IP address) Port id: 0060.b9xx.xxxx (Terminal MAC address) Port Description: LAN Port System Name: NEC IP Phone System Description: DT700 Series Time remaining: xx seconds System Capabilities: Bridge, Telephone Enabled Capabilities: Bridge, Telephone Auto Negotiation:...
Page 89 Part 2: VoIP Manual Location Identification Extended Power Inventory Hardware Revision WS-C3560-8PC Software Revision 12.2(55) SE Manufacturer Name Cisco Systems Model Name WS-C3560-8PC Network Policy (Voice) (VLAN ID, CoS, DSCP etc.) Network Policy (Voice Signal) (VLAN ID, CoS, DSCP etc.) Extended Power Via MDI Power priority Location ID...
Page 90 used in case the DHCP server is unavailable. Backup IP. This is where the terminal saves the network information that it receives from the DHCP server and uses it when the DHCP server is unavailable. If Spare IP is used the network information that can be preconfigured is: IP Address Default Gateway Subnet Mask...
Page 91: Sip Mlt Qos
Part 2: VoIP Manual Firmware Spare/Backup IP is supported on DT700 firmware V4.0.0.0 onwards Conditions/Comments If using a spare or backup IP address, the same address might be used by a different device making it impossible to communicate with the SV9100. If using a spare or backup IP address, Auto configuration cannot be executed.
Page 92 VLAN Settings It is possible for the SIP MLT to have specific VLAN and Priority settings. It is also possible for the PC uplink connection to have different VLAN and Priority settings. The VLAN and Priority settings must be made by programming each individual handset. Setting VLAN Values for the SIP MLT Press Menu, then 0 (Config) to enter the terminal program mode.
Page 93: Sip Mlt Ip Phone Manager
Part 2: VoIP Manual At the Login screen, enter the user name (default = ADMIN) and password (default = 6633222) and press the OK softkey. Press 1 for Network Settings. Press 6 for Advanced Settings. Press 4 for Type Of Service. Press 1 for RTP.
Page 94 using. IP Phone Manager now opens on your PC. SEARCHING FOR TERMINALS There are three methods to search for active terminals on your network.
Page 95 Part 2: VoIP Manual 1. Search The IP Phone manager sends a broadcast over the network in search of terminals. Active terminals respond to this broadcast with terminal information. There are three settings that change the Search Frequency and timing of the IP Phone Manager broadcast. 2.
Page 96 Terminal Connection After terminals are discovered by the IP Phone Manger Search functionality, they must be connected to before any action can take place. For terminals that need maintenance or further information communicated between them, select the check box and press Connect. When the Status Field indicates OK, the terminal is in active communication with IP Phone Manager.
Page 97 Part 2: VoIP Manual IP Phone Manager Commands SwitchPortCtrl Switch Port Control can enable or disable the PC Port on the connected IP Terminal(s). Reset This function resets the terminal(s) connected to the IP Phone Manager. Two options for resetting the Connected terminal(s) are available: Soft Reset –...
Page 98 This feature downloads various file types via a FTP/TFTP server. Select the server type to be used for downloads and the parameters that are required (IP Address of server, authentication name and passwords). Download Option: Use the Download Option Field to select the Terminal File type and enter a File name as required.
Page 99 Part 2: VoIP Manual Web Config The IP terminal has an HTTP server for web programming. Selecting this button starts a session with Internet Explorer (or the default web browser installed on the local PC) for all the connected and selected terminals. You have one browser session started for every selected terminal – this feature is used on an individual terminal.
Page 100: Sip Mlt Firmware Upgrade
Firmware Upgrade Below is an example of how to use IP Phone Manager to upgrade the firmware of SIP MLT’s Conditions The terminals must be operational. The IPPhoneManager must be located in the same network segment the terminals are in (only if using the Search function).
Page 101 Part 2: VoIP Manual Upgrade using the SIP MLT Menu Before following the upgrade procedure, the firmware files must be loaded onto a TFTP/FTP server. There are three firmware files depending on the type of handset being upgraded: - itlisipe.tgz (for use with ITL-2E/6E handsets) itlisipv.tgz (for use with ITL-8/12/24/32D handsets)
Page 102: Sip Mlt Auto Configuration
version information) Reboot the SIP MLT to start the upgrade process. SIP MLT Hardware Versions: ITL-2E/6E – 09.01.03.00 ITL-8/12/24/32D – 09.01.03.03 ITL-320C – 09.01.03.04 WARNING: If the firmware version set in PRG90-42-01 is different to the actual version loaded onto the TFTP/FTP server, the download by the SIP MLT will be repeated in an endless loop.
Page 103 Part 2: VoIP Manual Click Terminal...
Page 104 Click 1 st Server Address Assign the 1 st Server Address using the GPZ-IPLE IP Address, Easy Edit – Advanced Items/VoIP/ General Settings/IP Addressing/CCPU IPL IP Network Setup (PRG10-12-09) Click OK...
Page 105 Part 2: VoIP Manual Click 1 st Server Port. Assign port 5080. Click OK. After the changes are made click Save. 10. When the Save window opens, click Save as...
Page 106 11. In the Save as...window, name the file xxx.gz Example: To name the file test, enter test.gz 12. Place this file in the FTP Server Note: With the above config each handset will try and download the config every time it is reset which may be undesirable.
Page 107 Part 2: VoIP Manual Configuring an FTP Server The file generated in the IP Phone Manager must be placed in an anonymous login folder. The FTP server must be configured with an anonymous login account. The configuration of the FTP server will vary depending of the FTP server software. Refer to the manufactures instructions for further details.
Page 108: Sip Mlt Factory Default
Click ADD, and provide the following information: Name = Auto Config File Name Data Type = String Code = 153 Click on ADD, and give the following information: Name = Download Protocol Data Type = Byte Code = 163 Click OK. Option 151 is for DT700 Economy/Value/Sophisticated terminals (ITL-2E,ITL-6DE,ITL- 8LDE,ITL-12PA,ITL-12D,ITL-24D,ITL-32D,ITL-320C) Option 152 is for DT700 Gigabit Colour/Grayscale terminals (ITL-12CG,ITL-12DG)
Page 109 Part 2: VoIP Manual 331# Default terminal Green Flashing green Line key 1 Save settings Result = Pass Green after a delay Result = Fail Line key 1 Reset terminal* * If default was successful then the terminal is reset with default values. If default failed then the terminal is reset with existing settings.
Page 110: Standard Sip Extensions
Standard SIP (also referred to as 3 rd Party SIP) is an industry standard protocol and therefore there are many manufacturers hardware and software based phones. As these phones are not developed by NEC, and are not designed specifically for use on the SV9100, they do not support majority of the features that you would find on an SV9100 Keytelephone.
Page 111 Please make sure that only the required SIP ports belong to IP Duplication Groups Especially please do not change SIP timers unless advised by NEC Technical Support Department. For example, changing PRG 10-33-01 (Registration Expiry Time) on the SV9100 is currently not necessary or recommended for IP DECT as it is naturally set at the correct default for present purposes.
Page 112 A randomly generated password is created per port on the PBX to enhance the security of the system and minimise the risk or rogue SIP devices registering to the system. These passwords can be changed to make easier to configure and the password entered in system programming is then required to be entered in to the SIP device.
Page 113: Sip Extension Features
Part 2: VoIP Manual License The SV9100 requires licensing to allow the registration of Standard SIP handsets. One license is required for each registration to the SV9100. If no license is available, the SIP extension will fail to register. Other licenses may be required for certain Standard SIP features.
Page 114 Leased Line (Tie Line) Door Phone (operation explained below) Hold/Transfer Normal Hold Park Hold Group Hold Station Park Hold Type of transfer service Call Forward – Immediate Call Forward – Both Ring Call Forward – No Answer Call Forward – Busy Call Forward –...
Page 115 A SIP extension can belong to a Doorphone ring group but the operation differs slightly from that of an SLT telephone. The operation has been tested using an NEC IP DECT handset: Doorphone call button is pressed...
Page 116 T.38 Fax Relay Standard SIP extensions now have the ability to communicate using the T.38 Fax Relay protocol. By default any Fax tones transmitted to or from a Standard SIP extension will be sent as audio tones in the speech path. It is possible that these tones can be misinterpreted by the receiving device because of packet loss or errors due to the codec compression.
Page 117 Video communication between a Standard SIP terminal and the UC Client Softphone is not possible. The SIP terminal must pass interoperability testing by NEC before it can be supported. The video codec is not supported by the GPZ-IPLE card. The video codec information from the received SDP message will be forwarded to the destination terminal.
Page 118 Call Waiting SIP Call Waiting is designed for use with NEC IP DECT, the operation of this feature cannot be guaranteed with other Standard SIP Terminals. It allows a busy IP DECT user to receive a second call indication by a beep in the ear and some display information.
Page 119 Part 2: VoIP Manual call. A Doorphone call cannot be answered as a second call. To answer the Doorphone the first call must be cleared then wait for the Doorphone call to ring the IP DECT handset and answer as normal.
Page 120 Caller ID Update after Transfer Caller ID Update after Transfer is designed for use with NEC IP DECT, the operation of this feature cannot be guaranteed with other Standard SIP Terminals. It allows the CLI displayed on an IP DECT handset to be updated to show the transferred party’s name or CLI after the call has been transferred.
Page 121 Part 2: VoIP Manual The conference size supported is three party conference. SV9100 Requirements The following information provides the feature requirements. Main Software SV9100 R1.0 software or higher License The system must be licensed for standard SIP terminals with a BE114054 (License 5111). Service Condition This feature does not use the conference resources on the system, the standard SIP extension operates the conference by combining two voices and sends it to the other conference member.
Page 122 In this sequence, at first, SIP1 is talking with SIP2. If SIP1 pushes the conference key during conversation, the call is held. After that, SIP1 makes a call to another terminal (SIP3) that SIP1 wants to join the conference. After SIP3 answers the call, SIP1 pushes the conference key again, and then the call between SIP1 and SIP2 is retrieved.
Page 123: Ip Dect
IP DECT IP DECT IP DECT is an NEC product that combines the functionality of traditional DECT with the flexibility of the Standard SIP protocol giving a robust and reliable wireless solution. The handsets use the traditional DECT protocol to communicate with the DECT Access Points (DAPs) and the DAPs use the Standard SIP Protocol to communicate with the SV9100.
Page 124: Networking
The AP200s DAPs can be powered by PoE 802.3af or by local mains power supply. The AP300, AP300c, AP300e, AP400, AP400c, AP400e, AP400s can be only only be powered by 802.3af. IP DECT handsets can only use the G.711 codec. The supported IP DECT terminals are C124, G355, G955, i755, M155 G266, G566, G966, ML440.
Page 125 Part 2: VoIP Manual NetLink provides a seamless connection, using an IP network, to join multiple SV9100 communication servers into what would appear to be a single communications server. With a unified numbering plan, users can access any extension in the network as if they were in the same location. NetLink differs from other networking protocols because all slots, trunk ports and station ports belong to, and are controlled by, the Primary system.
Page 126 For the RTP and RTCP ports enter the starting port number for each IP address. Each DSP requires one RTP port and one RTCP port. Under normal circumstances there is no need to change these settings from default. NetLink Systems Easy Edit –...
Page 127 Part 2: VoIP Manual Each CODEC has different voice quality and compression properties. The correct choice of CODEC will be based on the amount of bandwidth available, the amount of calls required and the voice quality required. Available Codecs for NetLink G.711 64Kbps codec MOS 4.4 G.722 64Kbps codec MOS 4.4 G.726 32Kbps codec MOS 4.2...
Page 128 10-12-01 192.168.0.10 10-12-01 192.168.0.10 10-12-01 192.168.0.10 10-12-02 255.255.0.0 10-12-02 255.255.0.0 10-12-02 255.255.0.0 10-12-03 172.16.0.1 10-12-03 172.17.0.1 10-12-03 172.18.0.1 10-12-09 172.16.0.10 10-12-09 172.17.0.10 10-12-09 172.18.0.10 10-12-10 255.255.0.0 10-12-10 255.255.0.0 10-12-10 255.255.0.0 51-01-01 51-01-01 51-01-01 51-01-03 51-01-03 51-01-03 51-04-01 172.16.0.10 51-04-01 172.16.0.10 51-04-01 172.16.0.10 51-05-02...
Page 129 Part 2: VoIP Manual 3 – 172.18.0.10 3 – 172.18.0.10 3 – 172.18.0.10 51-05-02 51-05-02 51-05-02 51-06-01 51-06-01 51-06-01 84-26-01 172.16.0.20 84-26-01 172.17.0.20 84-26-01 172.18.0.20 Be aware that the above operation is not recommended as it is possible for the Primary System to move from one node to another if the Primary is reset.
Page 130 Asynchronous Settings The following programming commands are not sent from the Primary System to the Secondary Systems when Database Replication occurs: PRG10-01, PRG10-02, PRG10-12, PRG10-13, PRG10-14, PRG10-15, PRG10-16, PRG10-45, PRG51-01, PRG90-01, PRG90-09. SRAM Database The information held in SRAM is not transferred to Secondary Systems. So if a Fail-Over situation occurs terminals may lose DND, Caller ID History, etc.
Page 131: Netlink Advanced Features
Part 2: VoIP Manual 2.6.1.1 NetLink Advanced Features Netlink Advanced Features Fail-Over An added feature of NetLink is the Fail-Over feature. If the Primary System is turned off or disconnected, without the Fail-Over feature, all communication servers would stop working. With Fail-Over, one of the Secondary Systems, depending on the SV9100 programming, will take over as the Primary System allowing the other linked nodes to continue functioning.
Page 132 DTMF Relay (RFC2833) By default any DTMF tones transmitted across NetLink will be sent as audio tones in the speech path. It is possible that these tones can be misinterpreted by the receiving device because of packet loss or errors due to the codec compression.
Page 133 Part 2: VoIP Manual 2. The packet size exceeds the minimum size 3. 120mS after the previous packet was sent Buffering operation can reduce the network load by grouping together smaller packets until the minimum size is exceeded, therefore sending a smaller quantity of larger packets. However implementing the Buffering operation can cause problems when sending DTMF digits via a NetLink connection.
Page 134: Netlink Dsp Zones
the transmitted data. DTMF digits are always sent immediately by adding dummy data to ensure the packet size exceeds the MSS. 2.6.1.2 NetLink DSP Zones Netlink DSP Zones Description A DSP provides format conversion from circuit switched networks (TDM) to packet switched networks (IP). Each voice channel from the circuit switched network is compressed and packetised for transmission over the packet network.
Page 135: Aspirenet
Part 2: VoIP Manual 2.6.2 AspireNet AspireNet The SV9100 uses the GPZ-IPLE card or PRI card to connect multiple systems together over a Data Communication IP Network (Intranet). AspireNet is used to provide telephony services between the SV9100 and other SV9100’s, SV8100's or Aspire systems. There can be up to 16 Nodes in an AspireNet Network.
Page 136 Easy Edit – Blade Configuration/ISDN Port Setup/PRT Port Setup ISDN Line Mode - Set to one of the three available NW mode as follows: 3 - NW Mode (Leased Line) 4 - NW Mode (Interconnected Line) (generally used if the connected systems are directly connected via a fixed cable) 5 - NW Mode (Interconnected Line), (Fixed Layer 1 Forced NT Mode) NW Mode (Leased Line)
Page 137 Part 2: VoIP Manual The commands below are to set up the local and remote extension number ranges for each system. Easy Edit – Advanced Items/VoIP/Networking/AspireNet CVM/AspireNet Numb ering Plan System Numbering Easy Edit – Advanced Items/VoIP/Networking/AspireNet CVM/AspireNet Numb ering Plan/System Numb ering (PRG11-01) Set the relevant digits for the remote systems extension range to the correct length using 1 st Dial Digit, 1 st and 2 nd Dial Digit and Dial Digit Length.
Page 138 AspireNet Incoming Trunk Access Route Easy Edit – Advanced Items/VoIP/Networking/AspireNet CVM/AspireNet Remote Trunk Access/AspireNet Incoming Trunk Access Route (PRG21-16) This controls which local Trunk Route an incoming AspireNet call has access to for outgoing trunk calls. Set the Trunk Route on a per AspireNet Networking System ID. AspireNet Busy Lamp Information Easy Edit –...
Page 139 Part 2: VoIP Manual DTMF Relay (RFC2833) By default any DTMF tones transmitted across K-CCIS will be sent as audio tones in the speech path. It is possible that these tones can be misinterpreted by the receiving device because of packet loss or errors due to the codec compression.
Page 140: Aspirenet Centralised Voice Mail
ID 1 = 172.17.0.10 ID 1 = 172.16.0.10 ID 1 = 172.16.0.10 10-27-01 10-27-01 10-27-01 ID 2 = 172.18.0.10 ID 2 = 172.18.0.10 ID 2 = 172.17.0.10 1 = 3 Digit Type 2 1 = 3 Digit Type 8 ID 1 1 = 3 Digit Type 8 ID 2 = 3 Digit Type 8 ID 1 2 = 3 Digit Type 2...
Page 141 Part 2: VoIP Manual provided by: VM8000 InMail UM8000 External Voicemail If the AspireNet network has a mix of SV9100 systems and Aspire systems the Centralised Voicemail must be provided by one of the Aspire systems using: AspireMail DMS FMSU VMSU External Voicemail Centralised Voice Mail Easy Edit Programming...
Page 142: Aspirenet Features
Department Group Pilot Group 64 = 0 Department Group Pilot None Number Number Centralised Voice Mail Centralised Voice Mail Pilot Pilot Centralised Voice Mail Centralised Voice Mail None Department Group Department Group Numbering Plan Dial 6XX = Extension Numbering Plan Dial 6XX = Networking System Access 2.6.2.2...
Page 143 Part 2: VoIP Manual Transfer Voice Mail, Centralised ARS/F-Route Digits dialled by a user can be sent to the F-Route tables and specified as a AspireNet number by entering the Networked node ID (Trunk Group 101-150 correspond to Network ID’s 1-50) as the target trunk group number, calls will be routed to the target system via the node ID specified.
Page 144 1. Press Speaker (or lift handset) + Dial 888. Also allowed are 848 (Call Forward Immediate), 843 (Call Forward Busy), 845 (Call Forward No Answer, 844 (Call Forward, Busy/No Answer, or 842 (Call Forward Both Ring). Press Call Forwarding key (SC 851: code 16). 2.
Page 145 Part 2: VoIP Manual Operation To activate Call Forwarding Off-Premise 1. At system phone, press Speaker + Dial Call Forward Service Code (848, 843, 844, 845). Press Call Forward key (SC 851: 10, 11, 13 or 12) At an SLT, lift handset Dial 848, 843, 844 or 845. 2.
Page 146 3. SPK (or hang up at SLT) if you dialled 888 in step 1. Your Call Forwarding (Station) Programmable Function Key goes out. Camp On With Camp On, an extension user may call a busy extension and wait in line (Camp-On) without hanging The call goes through when the busy extension becomes free.
Page 147 Part 2: VoIP Manual Specified Trunk Group Access (804 + Trunk Group number). Operation The operation is automatic, the user dials the trunk access code in the normal way. Abbreviated Dial numbers will follow the trunk routing if set to TRG 0. For AspireNet ensure that the VOIPU ‘trunk’...
Page 148 in operation. In a single system, an extension within the same system can transfer a call to a Department Group and the call will ring an extension within the Department Group once the transferring user hangs up. In a networked system, the transfer will not go through and the call will recall the extension performing the transfer.
Page 149 Part 2: VoIP Manual trunk side. The Networking feature allows the following DISA Class of Service Options. Trunk Route Access Operator access External Paging The following are not available. Trunk Group Access Common Abbreviated dialling Internal Paging Specified trunk access Forced trunk disconnect Call Forward setting Break In...
Page 150 Operation To Set a Function Key to your Hotline partner: 1. Press Speaker to go off hook. 2. Dial service code 851. 3. Press Hotline key + partner’s extension number + HOLD. To place a call to your Hotline partner: 1.
Page 151 Part 2: VoIP Manual end, if the message is not sent (for example if the Ethernet cables are unplugged from the system) then the keep alive operation will not take place. When the keep alive operation occurs the link will be taken out of service: Any calls that are in progress will be released.
Page 152 Press Message Waiting key (SC 851: 38). At single line telephones, lift the handset and dial 841. Normally, your MW LED goes out. If it continues to flash, you may have new messages in your “Voice Mail” mailbox or a new “General Message”. Operator, Centralised It is possible to have a centralised network operator extension that can be dialled with the operator access code (0).
Page 153 Part 2: VoIP Manual 4. Make announcement to the networked system. 5. Press SPK to hang up. Park Park places a call in a waiting state (called a Park Orbit) so that an extension user may pick it up. Any extension user who is in the same Park Group as the extension which placed the call in Park can answer the call.
Page 154 1. Lift handset. If you want to place a trunk call, press a line key before lifting the handset. Depending on the setting of your ringdown timer, you may be able to dial an Intercom call before your ringdown goes through. If the destination has Handsfree Answerback enabled, your call will voice announce.
Page 155 Part 2: VoIP Manual Transfer The following types of Transfer are available with Networking: Screened Transfer Unscreened Transfer Transfer to busy extension Operation Transferring Trunk Calls To Transfer a trunk call to a co-worker’s extension: 1. At system phone or 2-Button telephone, press HOLD. At a single line telephone, hook flash.
Page 156: Sip Trunking
IP Trunk License(s) Networking Mode This mode is usually used to connect to other NEC systems (SV9100, SV8100, Aspire, XN120) using SIP although some SIP Carriers use this mode to provide external SIP Trunks. The systems connect to each other using an internal routing table. The call is always attempted even if the remote end point is down.
Page 157 Part 2: VoIP Manual the router using NAPT Router IP Address on either of the pages below depending on the mode being used. Networking Mode - Easy Edit – Advanced Items/VoIP/SIP Networking/Profile x/SIP Profile x - Networking Mode/SIP Profile x - Server Setup (PRG10-12-07,10-29-21) Carrier Mode (IP Address) - Easy Edit –...
Page 158 • SIP Multi Profiles must be configured with unique SIP Port numbers per profile. i.e. Profile 1 could use the default SIP port 5060 and Profile 2 could be configured to use 5061 etc. • SIP Multi Profile carrier configurations must be reachable through the same IP gateway.
Page 159 Part 2: VoIP Manual System A Configuration IP Settings Name Data 10-12 CCPU Network Setup Default Gateway = 172.16.10.254 NAT Route = 1(Yes) Default Gateway(WAN) = 10.1.1.254 IP Address (VoIP) = 172.16.10.10 Subnet Mask (VoIP) = 255.255.0.0 84-26 VoIPDB Basic Setup Slot #1: IP Address = 172.16.10.20 System Numbering Plan...
Page 160 10-29 SIP Server Information - SIP Profile 1 Setup Default Proxy = 1(On) Register Mode = 1(Manual) Domain Name = xxx.yyy.zzz.ne.jp Carrier Choices = 1(Carrier A) - SIP Profile 2 Default Proxy Port Number = 5062 Carrier Choices = 0(Standard) 10-36 SIP Registration - SIP Profile 1, Register ID 0 Registration = 1(On)
Page 161 Part 2: VoIP Manual Name Data 10-23 IP System Interconnection - System No: 1 Setup System Interconnection = 1 IP Address = 172.16.10.10 Dial = 1 - SIP Profile 1, Register ID 0 Registration = 1(On) 10-36 SIP Registration Information Setup User ID = 200 10-29 SIP Server Information...
Page 162 Each CODEC has different voice quality and compression properties. The correct choice of CODEC will be based on the amount of bandwidth available, the amount of calls required and the voice quality required. Available Codecs for SIP Trunks · G.711 64Kbps codec MOS 4.4 ·...
Page 163 Part 2: VoIP Manual Peer to Peer Video Support over SIP Trunk Peer to Peer video calls via SIP Trunk Interconnection are available for SIP video devices. Additionally, SIP trunk interconnection allows the SIP video device to access a Multi-Point Control Unit (MCU) to provide video conferencing over multiple SV9100 systems.
Page 164 3=Profile 4 4=Profile 5 5=Profile 6 10-29-21 NAT Router 0=Not Used This item should be disabled. 1=Used (default = 0) 15-05-50 Peer to Peer 0=Off This item should be enabled. mode 1=On (default = 0) 10-26-05 SIP CTI mode 0=disable This item should be disabled.
Page 165 Video communication between a Standard SIP terminal and the UC Client Softphone is not possible. o The SIP terminal must pass interoperability testing by NEC before it can be supported. o The video codec is not supported by the GPZ-IPLE card. The video codec information from the received SDP message will be forwarded to the destination terminal.
Page 166 Netlink Multiple SIP Carriers Description This allows the systems to register or connect to several different SIP Carriers or SIP servers at different locations. This means that SIP trunks can be used in multiple locations. This also means that the DSP consumption can be reduced when a SIP trunk is used by an extension on a Secondary system.
Page 167 Part 2: VoIP Manual 10-12 10-19 10-23 10-26 10-28 10-29 10-36 10-37 10-68 14-12 14-18 15-16 21-17 21-19 84-09 84-10 84-13 84-14 84-16 84-26 84-27 84-33 84-34 90-10 Note: Certain command settings may be removed when NetLink is first configured if SIP trunks are already enabled.
Page 168 Example Configuration The following programming details use the diagram below for SIP Trunk information (NetLink programming is not listed)
Page 169 Part 2: VoIP Manual Primary System Secondary System 10-12-01 192.168.0.10 10-12-01 192.168.0.10 10-12-02 255.255.0.0 10-12-02 255.255.0.0 10-12-03 172.16.0.1 10-12-03 172.16.1.1 10-12-07 20.20.20.1 10-12-07 20.20.20.2 10-12-09 172.16.0.10 10-12-09 172.16.1.10 10-12-10 255.255.0.0 10-12-10 255.255.0.0 10-28-01 10-28-01 10-28-02 aaa.sip 10-28-02 bbb.sip 10-28-05 Domain name 10-28-05 Domain name 10-29-03...
Page 170 10-68-01 10-68-01 10-68-02 10-68-02 10-68-03 10-68-03 14-18-05 Profile 1 14-18-05 Profile 1 84-26-01 172.16.0.20 84-26-01 172.16.1.20 SIP Register ID 32 Register ID’s can be programmed on each NetLink system that has SIP Trunks configured. Register ID 0 to 31 uses the User ID’s programmed in PRG10-36. Trunks SIP Register ID’s for trunks can only be applied to trunks that belong to the same system as the Register Stations...
Page 171 Part 2: VoIP Manual Register Error Format Terminal LCD Example NetLink ###:SIP(XX) REG FAILED($$$) ###:Alarm number (60) “60:SIP(07) REG FAILED(403) ” Disabled XX:Register ID (1-32) $$$:Err code (000-999) NetLink ###:SIP(XX) REG NG($$$) -% “60:SIP(07) REG NG(403) -02 ” %% System ID (1-16) Enabled Register Authentication Error Format...
Page 172 To header: To: sip:[email protected] From header: From: sip:[email protected] Request-URI: Invite sip:[email protected] SIP/2.0 No Setting To header: To: sip:[email protected] From header: From: sip:[email protected] Request-URI: Invite sip:[email protected] SIP/2.0 To header: To: sip:[email protected] From header: From: sip:[email protected] Request-URI: Invite sip:[email protected] SIP/2.0 No Setting To header: To: sip:[email protected] From header: From: sip:[email protected] When PRG84-14-13 is set to International Access Mode for any out bound call via SIP trunk if the leading...
Page 173 Part 2: VoIP Manual When ‘+’ is used in the incoming call from an external 0:Off 0:Off SIP carrier, then delete ‘+’ only. When a call comes in with a ‘+’ and Country Code other than what is defined in PRG10-02-01 then delete ‘+’ 0:Off 1:On only.
Page 174 ‘+’ and Country Code as assigned in PRG10-02-01 then delete the ‘+’ and Country Code and do not add the International Access Code in PRG10-02-02. Incoming number dialled: +4902131795770 PRG84-14-16 = 1 PRG10-02-02 = 00 Resulting number displayed on terminal history of incoming calls PRG10-02-01 = 0 PRG10-02-01 = 49 When a call comes in with a ‘+’...
Page 175: Sip Trunks - Carrier Mode
Part 2: VoIP Manual Enable VBD on SIP Trunk by using the ‘Voice Band Data (VBD)’ option in Easy Edit - Advanced Items/VoIP/ SIP Networking/Profile x/SIP Profile x - Trunk General Settings/SIP Profile x - CODEC Settings (PRG84-13- 66). For each VBD device connected to the system set the Terminal Type to SPECIAL using the ‘Terminal Type’...
Page 176 used for the outgoing and incoming calls. Calls may be restricted only to/from the specified address. (PRG10-29-01/02) Default Proxy Address – This is the IP address of the SIP Server where the outgoing calls will be routed (PRG10-29-03) Default Proxy Port – This is the UDP port number that is used on the ITSP’s SIP Server. (PRG10-29-04) Register Mode –...
Page 177 Part 2: VoIP Manual x - E.164 Numb er Formatting/SIP Profile x - E.164 Numb er Format Settings Incoming/Outgoing SIP Trunk for E.164 - Defines the settings for support of E.164 number formatting when required for specified SIP carriers. (PRG84-14-13) Enabled –...
Page 178 DNS Source Port – This is the source port number of the customer’s (or ISP’s) DNS server. (PRG10-29- Register Sub Mode - Prevents an invalid INVITE message. If the register information that SV9100 send to SIP server and the Invite information that SV9100 receives are different, SV9100 sends “404 Not Found” Message.
Page 179 Carrier Mode IP Address Command Entry 10-12-01 192.168.0.10 10-12-02 255.255.0.0 10-12-03 172.16.0.1 10-12-07 10-12-09 172.16.0.10 10-12-10 255.255.0.0 10-28-01 nec.co.uk 10-28-02 SV9100 10-28-05 IP Address 10-29-01 Enabled 10-29-02 Enabled 10-29-03 10-29-05 Manual 10-29-06 10-29-21...
Page 180 10-36-01 Enabled 10-36-02 10-36-03 10-36-04 10-68-01 10-68-02 10-68-03 4 ports 14-18-05 Profile 1 84-26 VoIP Gateway = 172.16.0.20 Carrier Mode DNS Command Entry 10-12-01 192.168.0.10 10-12-02 255.255.0.0 10-12-03 172.16.0.1...
Page 181: Sip Trunks - Networking Mode
Part 2: VoIP Manual 10-29-09 10-29-11 10-29-12 10-29-13 10-29-21 Used 10-36-01 Enabled 10-36-02 10-36-03 10-36-04 10-68-01 10-68-02 10-68-03...
Page 182 Trunk Type - Set to SIP (PRG10-68-01) Start Port - Enable the logical starting port number (PRG10-68-02) Number of Ports – Enter the required number of SIP trunk ports (PRG10-68-03) Assign the SIP Profile used to the SIP trunks assigned in Easy Edit –...
Page 183 Part 2: VoIP Manual 10-12- 172.16.0.1 10-12-03 172.17.0.1 10-12-03 172.18.0.1 10-12- 172.16.0.10 10-12-09 172.17.0.10 10-12-09 172.18.0.10 10-12- 255.255.0.0 10-12-10 255.255.0.0 10-12-10 255.255.0.0 10-23- System 1 = Enabled 10-23-01 System 1 = Enabled 10-23-01 System 1 = Enabled System 2 = Enabled System 2 = Enabled System 2 = Enabled 10-23-...
Page 184: K-Ccis
Data Communication IP Network (Intranet). Key-Common Channel Interoffice Signalling (K-CCIS) is used to provide telephony services between the SV9100 and another SV9100 or SV8100. The SV9100 uses the NEC proprietary CCIS Peer to Peer protocol over IP to communicate between systems.
Page 185 Part 2: VoIP Manual Resources per system. Each system in the network must have: GCD-CP10 GPZ-IPLE Feature Networking (K-CCIS) license(s): BE114066 K-CCIS Easy Edit Programming Easy Edit – Advanced Items/VoIP/Networking/K-CCIS K-CCIS Setup Easy Edit – Advanced Items/VoIP/Networking/K-CCIS/K-CCIS Setup/K-CCIS IP Trunk Assignment Trunk Type - Set to CCIS (PRG10-68-01) Start Port - Enable the logical starting port number...
Page 186 · G.722 64Kbps codec MOS 4.4 · G.726 32Kbps codec MOS 4.2 · G.729 8Kbps codec MOS 4.0 The bandwidth values quoted for these codec’s are for the digitized speech in one direction only. The actual bandwidth required for a call will depend on many other factors and will be much higher than these figures.
Page 187 Part 2: VoIP Manual Destinations (PRG50-11) Enter the Destination Point code of the system to be controlled. Up to sixteen destination point codes can be entered Easy Edit – Advanced Items/VoIP/Networking/K-CCIS/K-CCIS Night Mode Switching/K-CCIS Night Switching Mode (PRG50-12) Enter the required mode settings for Day Mode and Night Mode K-CCIS Misc Easy Edit –...
Page 188 System A System B System C 10-12- 192.168.0.10 10-12- 192.168.0.10 10-12- 192.168.0.10 10-12- 255.255.0.0 10-12- 255.255.0.0 10-12- 255.255.0.0 10-12- 172.16.0.1 10-12- 172.17.0.1 10-12- 172.18.0.1 10-12- 172.16.0.10 10-12- 172.17.0.10 10-12- 172.18.0.10 10-12- 255.255.0.0 10-12- 255.255.0.0 10-12- 255.255.0.0 10-40- Enable 10-40- Enable 10-40- Enable 10-40-...
Page 189: K-Ccis Features
Part 2: VoIP Manual Licensing Each SV9100 needs to be licensed to use IP Trunks for K-CCIS. One license gives that system the ability to use one IP Trunk. Each K-CCIS node requires the relevant number of IP Trunk licenses. K-CCIS IP Trunk Availability/Licensing Limitation There is a limitation when configuring K-CCIS Trunk Availability that the correct number of system port licenses must be free and available on the SV9100.
Page 190 If PRG 34-07-05 is left at default (30) the transferred call recalls to the station that performed the transfer when not answered. A UNIVERGE SV9100 station can receive a K-CCIS transferred call as a camp-on call if allowed by Class of Service.
Page 191 Part 2: VoIP Manual Trunk-to-Trunk Transfer must be allowed in Program 14-01-13 (Trunk-to-Trunk Transfer Yes/No Selection). A blind transfer across a K-CCIS link cannot be completed until ringback tone is received at the transferring station. Related Feature List Link Reconnect – K-CCIS Station-to-Station Calling –...
Page 192 Guide to Feature Programming This guide provides a list of associated Programs that support this feature. Call Back – K-CCIS Feature Description This feature allows a station to set a Call Back after dialling across K-CCIS to a busy destination. The station that set the Call Back will receive a call as soon as the busy station becomes idle.
Page 193 The activation and cancellation of this feature may be accomplished by either the station user or an Attendant position if allowed by Class of Service (COS). Attendant Positions can be used to cancel Call Forward – All Call system-wide. For more details, refer to the UNIVERGE SV9100 Features and Specifications Manual. System Availability All multiline terminals...
Page 194 Restore handset or press Speaker. To set Call Forward – All Calls – K-CCIS from a Multiline Telephone (Open Numbering Plan): Press the Call Forward – All ON/OFF key. Dial 1 to set. Dial the trunk Access Code. Dial the Office Code number. Dial the distant K-CCIS station number.
Page 195 Attendant, in another office in the K-CCIS network. The activation and cancellation of this feature may be accomplished by either the station user or an Attendant position, if allowed by Class of Service (COS). For more details, refer to the UNIVERGE SV9100 Features and Specifications Manual. System Availability...
Page 196 Required Components: GPZ-IPLE Operating Procedures To set Call Forward – Busy/No Answer – K-CCIS from a Multiline Telephone (Closed Numbering Plan): Press the Call Forward – Busy/No Answer ON/OFF key. Dial 1 to set. Dial the remote K-CCIS station number. Press Speaker.
Page 197 Part 2: VoIP Manual Call Forward Split Internal/External is not supported. Forwarding to Voice Mail is not included in the Maximum Hop Count. Call Forward continues to operate to a MLT that has been removed. Related Feature List Call Forwarding – Busy/No Answer – K-CCIS Multiple Call Forwarding –...
Page 198 When two or more stations attempt to retrieve the parked call, only one station can retrieve the call. A station connected to a PBX can retrieve a parked call in an UNIVERGE SV9100, but the station connected to the UNIVERGE SV9100 system cannot retrieve a parked call in a PBX.
Page 199 Part 2: VoIP Manual Guide to Feature Programming This guide provides a list of associated Programs that support this feature. Park Originate System Remote System (Call Park Retrieve)
Page 200 This feature allows a station user to transfer incoming or outgoing Central Office, intraoffice and interoffice calls to another station in the K-CCIS network without Attendant assistance. For more details, refer to the UNIVERGE SV9100 Features and Specifications Manual. System Availability...
Page 201 Restore the handset (transfer is completed). Service Conditions General: A UNIVERGE SV9100 station can receive a K-CCIS transferred call as a camp-on call if allowed by Class of Service. Restrictions: Trunk-to-Trunk Transfer must be allowed in Program 14-01-13 (Trunk-to-Trunk Transfer Yes/No Selection).
Page 202 Feature Description This feature permits the station name of a calling or called party at another switching office to be displayed on a multiline terminal, through the K-CCIS network. For more details, refer to the UNIVERGE SV9100 Features and Specifications Manual.
Page 203 For incoming or outgoing K-CCIS calls, the Calling/Called Name and Number are displayed for the entire length of the call including the Elapsed Call Time. RESTRICTIONS: In the UNIVERGE SV9100 system, only 12 digits/characters can be entered for each station name. Related Feature List Calling Number Display – K-CCIS Station-to-Station Calling –...
Page 204 Normal call handling procedures apply. Service Conditions General: Both the caller/calling station number and name can be displayed on an UNIVERGE SV9100 station if allowed by Class of Service. For incoming or outgoing K-CCIS calls, the Calling/Called Name and Number are displayed for the entire length of the call including the Elapsed Call Time.
Page 205 Part 2: VoIP Manual Calling Party Number (CPN) Presentation from Station – K-CCIS Feature Description Calling Party Number (CPN) Presentation from Station K-CCIS feature allows each station of the remote systems a unique 10-digit number (the DID number of the originating station) to be sent out over the PRI circuit of the main system.
Page 206 Centralised DSS/BLF key allows direct access to the station through the K-CCIS network. Do Not Disturb and Voice Mail Message Waiting on Line key indications are also supported. For more details, refer to the UNIVERGE SV9100 Features and Specifications Manual. System Availability...
Page 207 The Voice Mail MSG Waiting has priority over any other state of the flashing line key or One-Touch key. Restrictions: This feature is not supported between UNIVERGE SV9100 and NEAX PBXs. This feature is supported with a Closed Numbering Plan only (not available with an Open Numbering plan).
Page 208 The BLF information is expelled when data cannot be sent if the K-CCIS link is down. The UNIVERGE SV9100 does not send BLF information again when the K-CCIS link is restored. BLF messages can be forwarded up to eight times in the network. When designing the K-CCIS network, this should be a consideration.
Page 209 Day/Night mode switching from an Attendant Position at the main office. When a UNIVERGE SV9100 system is connected to another UNIVERGE SV9100 system, the main office can control remote offices. For more details, refer to the UNIVERGE SV9100 Features and Specifications...
Page 210 The LED for any Feature Access key assigned for Night Mode transfer and the Night Mode key on the Attendant console are On. If the K-CCIS link is not available due to network trouble, the UNIVERGE SV9100 main office resends the K-CCIS Day/Night Mode switch command every 16 minutes.
Page 211 Part 2: VoIP Manual Program 50-03-01 (Destination Point Code Transfer Assignment) must be set for all offices for the Centralised Day/Night Mode feature. Related Feature List Assigned Night Answer (ANA) Authorisation Code Automatic Day/Night Mode Switching Centralised Billing – K-CCIS Code Restriction Dial Access to Attendant –...
Page 212 For Night Transfer Feature...
Page 213 Part 2: VoIP Manual Centralised E911 – K-CCIS Feature Description This feature allows a remote system to transmit a Calling Party Number to the 911 Emergency System over a K-CCIS direct or tandem connection. System Availability Terminal Type: All Stations Required Components GPZ-IPLE Operating Procedures...
Page 214 Dial Access to Attendant – K-CCIS Feature Description This feature allows a station user to call an Attendant by dialling a call code through the K-CCIS network. For more details, refer to the UNIVERGE SV9100 Features and Specifications Manual. System Availability Terminal Type:...
Page 215 Shows Office Code if using Open Numbering Plan. If using an Open Numbering Plan and a call is made to an UNIVERGE SV9100 Attendant Position, the operator office code is included with the name. When making a call from a UNIVERGE SV9100 Attendant Position across a K-CCIS network, the Caller ID Name and Number display is the same as for a station-to-station call.
Page 216 For Remote System...
Page 217 This feature is supported when a Closed Numbering Plan or Open Numbering is used. The UNIVERGE SV9100 system supports DID Digit Conversion when using station numbers with 2 to 8 digits. An extension on a remote system can be the destination for the DID Received Vacant Number Operation Assignment (Program 22-09-02).
Page 218 This feature allows two connected Multiline Telephones to be placed on hold simultaneously over the K- CCIS link. This enables the held parties to answer or originate a call from a secondary line or intercom path. For more details, refer to the UNIVERGE SV9100 Features and Specifications Manual. System Availability...
Page 219 Part 2: VoIP Manual Guide to Feature Programming This guide provides a list of associated Programs that support this feature. Elapsed Time Display – K-CCIS Feature Description This feature provides an Elapsed Call Time on the LCD which shows the duration of time that a multiline terminal is connected to any call through the K-CCIS network.
Page 220 1,000's group for 6-digit station numbers, and 10,000's group for 7-digit station numbers. Example: Feature Description This feature allows telephone numbers to be assigned to any stations in the K-CCIS network, based solely upon numbering plan limitations. For more details, refer to the UNIVERGE SV9100 Features and Specifications Manual.
Page 221 Tenant service is not provided, i.e., numbers cannot be duplicated for different tenants. Extension numbers should not start with 0, 9, * or #. For non-K-CCIS feature support, refer to the UNIVERGE SV9100 Features and Specifications Manual, Flexible Numbering Plan feature.
Page 222 Handsfree Answerback – K-CCIS Feature Description This feature allows Multiline Telephone station users to respond to voice calls through a K-CCIS network without lifting the handset. For more details, refer to the UNIVERGE SV9100 Features and Specifications Manual. System Availability...
Page 223 Hot Line – K-CCIS Feature Description This feature allows two stations at different nodes in the K-CCIS network to be mutually associated on automatic ringdown through the K-CCIS network. For more details, refer to the UNIVERGE SV9100 Features and Specifications Manual. System Availability...
Page 224 Operating Procedures To execute at any station programmed for Hot Line: Lift the handset or press Speaker. The remote K-CCIS station is called. Service Conditions General: Any multiline terminal (a maximum number of 512 stations) can be assigned for Hot Line – (K-CCIS). Either multiline terminal in a Hot Line –...
Page 225 Part 2: VoIP Manual Link Reconnect – K-CCIS Feature Description This feature provides the system that is connected to a K-CCIS network with the ability to release the redundant K-CCIS link connections and reconnect the link with the system for efficient usage of the K-CCIS trunks.
Page 226 Example: A trunk call (CO/PBX/TIE/DID/K-CCIS) over a K-CCIS network is transferred or forwarded to another station or trunk within the same office as the original incoming trunk. Call Forward to Trunk Line...
Page 227 Part 2: VoIP Manual Link reconnect occurs after answering a transferred or forwarded K-CCIS call. Restrictions: Answer supervision is required for Link Reconnect to occur. For outgoing calls on analogue trunks, Answer supervision is based on the Elapsed Call Time - Program 21-01-03 (Trunk Interdigit Time).
Page 228 Feature Description This feature allows a Multiple Call Forwarding – All Calls sequence to be forwarded over a K-CCIS network to a station in another office. For more details, refer to the UNIVERGE SV9100 Features and Specifications Manual. System Availability...
Page 229 Part 2: VoIP Manual Dial 0 to cancel. Press Speaker. - OR - Lift the handset or press Speaker. Dial Access Code 741 (set as default), and Dial 0 to cancel. Restore handset or press Speaker. Service Conditions General: Multiple Call Forwarding – All Calls – K-CCIS can forward a call up to seven times across K-CCIS links (up to seven hops) depending on system data.
Page 230 Related Feature List Call Transfer – All Calls – K-CCIS Call Forwarding – All Calls – K-CCIS Call Forwarding – Busy/No Answer – K-CCIS Multiple Call Forwarding – Busy/No Answer – K-CCIS Link Reconnect – K-CCIS Uniform Numbering Plan – K-CCIS Guide to Feature Programming This guide provides a list of associated Programs that support this feature.
Page 231 Feature Description This feature allows a Multiple Call Forwarding – Busy/No Answer sequence to be forwarded over a K-CCIS network to a station in another office. For more details, refer to the UNIVERGE SV9100 Features and Specifications Manual. System Availability...
Page 232 Operating Procedures To set Call Forward – Busy/No Answer - K-CCIS from a Multiline Telephone (Closed Numbering Plan): Press the Call Forward – Busy/No Answer ON/OFF key. Dial 1 to set. Dial the remote K-CCIS station number. Press Speaker. - OR - Lift the handset or press Speaker.
Page 233 Part 2: VoIP Manual Service Conditions General: Multiple Call Forwarding – Busy/No Answer Calls - K-CCIS can forward a call up to five times across K-CCIS links (up to five hops) depending on systems data. Multiple Call Forwarding over a K-CCIS link is combined with Multiple Call Forwarding – All Calls/ Busy/No Answer.
Page 234 Restrictions: Trunk-to-Trunk Transfer must be allowed in Program 20-11-14 [Class of Service Options (Hold/ Transfer Service) Trunk-to-Trunk Transfer Restriction]. Related Feature List Call Forwarding – All Calls – K-CCIS Call Forwarding – Busy/No Answer – K-CCIS Multiple Call Forwarding – All Calls – K-CCIS Link Reconnect –...
Page 235 This feature allows users to access internal or external paging from remote sites across the K-CCIS network. Local stations where the external paging equipment is installed can use the Meet-Me Answer feature to answer the page and establish a station-to-station K-CCIS call. For more details, refer to the UNIVERGE SV9100 Features and Specifications Manual.
Page 236 Program 31-02-02 (Internal All Call Paging Receiving) applies to Paging Access – (K-CCIS). Restrictions: Amplifiers and speakers must be locally provided. Combined Paging is not supported over K-CCIS. Internal Paging across K-CCIS is supported only between UNIVERGE SV9100 and UNIVERGE SV9100. Related Feature List Background Music Over External Speakers...
Page 237 Part 2: VoIP Manual For Paging Installation...
Page 239 Part 2: VoIP Manual For Remote System...
Page 240 Quick Transfer to Voice Mail – K-CCIS Feature Description A station user transferring a call can force the call to be transferred to the called party voice mail box after the transferred call recalls, after an internal station number is dialled while performing a screened transfer, or during intercom calls.
Page 241 Station-to-Station Calling – K-CCIS Feature Description This feature permits any multiline terminal user to dial another multiline terminal directly through a K-CCIS network. For more details, refer to the UNIVERGE SV9100 Features and Specifications Manual. System Availability Terminal Type: All multiline terminals...
Page 242 In the second plan, the station user dials a one-, two- or three-digit office code and a telephone number from two to eight digits. For more details, refer to the UNIVERGE SV9100 Features and Specifications Manual. System Availability...
Page 243 Part 2: VoIP Manual Lift the handset or press Speaker. Dial the remote K-CCIS station number. To call a station at another office using Numbering Plan 2 (Open Numbering Plan): Lift the handset or press Speaker. Dial the trunk Access Code. Dial the Office Code number.
Page 244 Service Conditions General: The UNIVERGE SV9100 can assign a Feature Access/One Touch Button as a Voice Call key. This performs the same operation as pressing 1. Any station in the same system can use Directed Call Pick Up to retrieve the Voice Call over K- CCIS.
Page 245 Part 2: VoIP Manual System Availability Terminal Type: All Stations REQUIRED COMPONENTS: VM8000 InMail GPZ-IPLE Operating Procedures To access voice mail from a Multiline Telephone in the Main system: Lift the handset or press Speaker. Dial pilot number for voice mail. When voice mail answers use soft keys to navigate.
Page 246 In a PBX to KTS network, Centralised Voice Mail is supported only via closed numbering plan. In a PBX to KTS network, Centralised Voice Mail is supported using the PBX voice mail. Extension numbers of remote extensions must be cleared from command 11-02. Single Line Telephones (SLTs) connected to the AP(A)-R Unit/AP(R)-R Unit or APA-U Unit/APR-U Unit cannot be used to transfer a Trunk call across the K-CCIS Network to another station or Voice Mail.
Page 247 Part 2: VoIP Manual...
Page 249 Part 2: VoIP Manual...
Page 250: Networking Dsp Usage
2.6.5 Networking DSP Usage Networking GPZ-IPLE DSP Usage NetLink GPZ-IPLE DSP Usage Peer to Peer is enabled where possible Primary System Secondary Secondary System System 1 DTer Trunk Trunk Trunk Trunk DT70 S1=1 S1=1 S2=1 S2=1 DT80 DTer P=2 S1=1 S1=1 P=2 S2=1 S2=1 Prima S1=1 S1=1...
Page 251 Part 2: VoIP Manual DT70 S1=2 S1=1 S1=1 S2=1 S2=1 DT80 Secon dary S1=1 S1=1 Syste P=1 S1=1 S1=1 S1=1 S1=1 S1=1 S2=1 S2=1 S1=1 S1=1 S1=1 S1=1 Trunk S1=1 S1=1 S1=1 S2=1 S2=1 DT70 S2=2 S1=1 S1=1 S2=1 S2=1 DT80 Secon dary...
Page 252 N1=1 N1=1 N1=2 N1=1 N1=1 Trunk N2=2 N2=1 N3=1 N3=1 DT700/ N2=2 N2=2 N2=2 N2=2 N2=2 N2=1 DT800 N1=2 N1=1 N3=2 N3=1 N2=1 N2=1 N2=1 N2=1 N2=1 N1=2 N1=1 N3=2 N3=1 N2=2 N2=2 N2=2 N2=2 N2=2 N2=1 Trunk N1=2 N1=1 N3=2 N3=1 N2=1...
Page 253 Part 2: VoIP Manual K-CCIS GPZ-IPLE DSP Usage Peer to Peer is enabled where possible Node 1 Node 2 Node 3 Trun Trunk Trun Trunk Trunk Trunk DT700/ N1=2 N1=1 N2=2 N2=1 N3=2 N3=1 DT800 N1=1 N1=1 N1=1 N1=1 N1=1 N2=2 N2=1 N3=2...
Page 254: Voip Troubleshooting
4. Click OK. A Command prompt window opens. 5. Type ping 192.168.1.200. The below screen shot shows that the UNIVERGE SV9100 system has replied to the Ping request – this indicates that the UNIVERGE SV9100 system is correctly connected to the network.
Page 255: Appendix A: Tcp_Udp Port Numbers
Part 2: VoIP Manual Depending on whether you have connected the GCD-CP10 or GPZ-IPLE to the network the following IP addresses can be checked. Description PRG Command Default Value GCD-CP10 IP address 10-12-01 192.168.0.10 GPZ-IPLE IP address 10-12-09 172.16.0.10 VoIP Media Gateway IP Address 84-26-01 172.16.0.20 Appendix A: TCP_UDP Port Numbers...
Page 256 K-CCIS Server Signalling 57000 K-CCIS Client Signalling 59000 DHCP Server 5964 H.323 H.245 5600 H.323 Trunk RAS 20001 H.323 Trunk Signalling 1720 SIP MLT Signalling (handset) 5060 SIP MLT RTP (handset) 3462 10020 (increment GPZ-IPLE RTP by two per call) 10021 (increment GPZ-IPLE RTCP by two per call)
Page 257: Appendix B: Sip Configuration Example
Part 2: VoIP Manual Appendix B: SIP Configuration Example Appendix B: SIP Configuration Example SIP Softphone Configuration (CounterPath, X-Lite) X-Lite is a software based SIP phone, available from CounterPath Corporation. The software is freely available from their website. X-Lite is a reduced feature version of their commercial software, eyeBeam. If G.729 CODEC is required it is necessary to use the eyeBeam software.
Page 258 Display Name: The name that will appear on the softphone display. User name: The extension number assigned to the extension port Password: The password assigned in Authentication Password in Easy Edit – Advanced Items/IP SIP/ SIP Terminal Options/SIP Terminal Setup. (PRG15-05-16) Authorization User name: Same as User name Domain: The IPLE card IP address and port number.
Page 259 Part 2: VoIP Manual Please refer to http://www.counterpath.com/ for further information on this product...
Page 260: Appendix C: Tos Field Values
2.10 Appendix C: ToS Field Values Appendix C: ToS Field Values...
Page 261 Part 2: VoIP Manual...
Page 262: Appendix D: Configuration Of External Dhcp Server
2.11 Appendix D: Configuration of External DHCP Server Configuration of External DHCP Server Note: - Although the SV9100 supports DHCP Server functionality, it is only designed for demonstration or test purposes. Please be aware that no support can be offered if the DHCP Server is used on a customer’s site.
Page 263 Part 2: VoIP Manual 5. Click OK and this will create a new Option type for the DHCP server. 6. Enter each octet of the SV9100 IPLE card IP address, click Add after each octet. The digit 1 must always precede the IP Address as shown above.
Page 264 9. Scroll down the Available Options list and tick 120 SIP Server, click OK. 10.The DHCP server is now ready to provide DT800/DT700 phones with the SIP Server address. The example below shows the necessary steps to add Option 168 (SIP Server Port) to a Windows 2003 Server.
Page 265 Part 2: VoIP Manual In the DHCP server, highlight the server machine on the left hand side. Right click on the server and select Define Vendor Classes. 2. Click Add and enter the information show below. The ID and Binary information is entered automatically when you enter the Display name, Description and ASCII value.
Page 266 3. Click OK. 4. Click Close. 5. Highlight the server again, then right click and select Set Predefined Options. 6. Change the Option class to NECDT700.
Page 267 Part 2: VoIP Manual 7. Click Add and enter the information show below. 8. Click OK. 9. In the Word box, enter 0x13d8 as shown below.
Page 268 10.Click OK. 11.Highlight scope options on the left hand side. Then right click and choose Configure Options. 12.Click Advanced and change the vendor class to NECDT700. 13.Place a check mark next to 168 SIP Port, check that 0x13d8 is in the Word box.
Page 269: Appendix E: Sv9100 Rfc Support
Part 2: VoIP Manual 14.Click OK. The DHCP server is now ready to provide IP Phones with the SV9100 SIP Port 5080. 2.12 Appendix E: SV9100 RFC Support Appendix E: SV9100 RFC Support SV9100 Support Title Comments Number Std. Trunk System Station User Datagram Protocol (UDP) THE TFTP PROTOCOL...
Page 270 INTERNET PROTOCOL INTERNET CONTROL MESSAGE PROTOCOL TRANSMISSION CONTROL PROTOCOL (TCP) An Ethernet Address Resolution Protocol (ARP) TELNET PROTOCOL FILE TRANSFER PROTOCOL (FTP) DOMAIN NAMES - CONCEPTS 1034 Resolve only supported AND FACILITIES (DNS) DOMAIN NAMES - 1035 IMPLEMENTATION AND SPECIFICATION (DNS) A Simple Network Management 1157 Protocol (SNMP)
Page 271 Part 2: VoIP Manual Lightweight Directory Access 2252 Protocol (v3): Attribute Syntax Definitions Lightweight Directory Access Protocol (v3): UTF-8 String 2253 Representation of Distinguished Names The String Representation of 2254 LDAP Search Filters 2255 The LDAP URL Format A Summary of the X.500(96) User 2256 Schema for use with LDAPv3 SDP: Session Description...
Page 272 Reliability of Provisional 3262 Responses in the Session Initiation Protocol Session Initiation Protocol (SIP): 3263 Locating SIP Servers An Offer/Answer Model with the 3264 Session Description Protocol (SDP) Session Initiation Protocol (SIP)- 3265 Specific Event Notification The Session Initiation Protocol 3311 (SIP) UPDATE Method A Privacy Mechanism for the...
Page 273 Part 2: VoIP Manual STUN – Simple Traversal of User Note: This RFC specification is 3489 Datagram Protocol (UDP) not supported and no Through NAT requirement from market yet. The Session Initiation Protocol 3515 (SIP) Refer Method RTP: A Transport Protocol for 3550 Real-Time Applications MIME Type Registration of RTP...
Page 274 The tel URI for Telephone 3966 Numbers Network News Transfer Protocol 3977 (NNTP) Session Timers in the Session 4028 Initiation Protocol (SIP) The Secure Shell (SSH) Protocol 4250 Assigned Numbers The Transport Layer Security MA4000/OW5000 can use 4346 (TLS) Protocol HTTPS using TLS...
Page 275 SV8100 Voice Over IP Manual...

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