1. When an OCS client logs in, the CS1000 is informed so it can route calls to that user.
2. An INVITE message is sent from the OCS client to the FE Server, then to the Application Proxy and NRS to route to the correct Signalling Server.
3. The Signalling Server processes the request and acquires the phone, responding with a 200 OK message back through the chain to the OCS client.
Microsoft lync server 2010 protocol workloads posterbigwalker
This document provides a summary of the Microsoft Lync network architecture and protocols. It includes diagrams showing the typical placement of internal and external firewalls, edge servers, directors, pools, and additional workload-specific servers. The document also lists common ports, protocols, and DNS records used across the Lync infrastructure.
SIP is an application layer protocol for initiating, modifying, and terminating multimedia sessions over IP networks. It uses URIs to address endpoints and can be used for voice, video, instant messaging and other applications. SIP includes components like user agents, servers, and gateways. Common SIP messages are INVITE to create a session, ACK, BYE, and REGISTER. SDP is used to describe multimedia sessions and carry information like media formats and ports. Common codecs are G.711 for voice and H.264 for video.
SPARSH VP248 is a high-definition VoIP phone built with superior acoustics and elegant design to provide unsurpassed audio quality and rich user experience.
Based on open-standard SIP protocol, SPARSH VP248 is interoperable with any standard SIP infrastructure such as IP-PBX, SIP Proxies, Softswitches and Stand-alone applications.
SPARSH VP248 is designed for power users, knowledge workers and managers for quick access totheadvance system features and functions. A feature-packed IP phoneenables user to work efficiently with advance call handling capabilities
Office Comunnications Server 2007 R2 PosterPaulo Freitas
This document provides an overview of the workload architecture for a unified communications solution. It includes a legend defining the various hardware components, diagrams showing traffic flows between internal and external systems, and details on DNS records, certificates, and firewall configurations required. The solution comprises multiple workloads including IM and presence, audio/video conferencing, enterprise voice, and application sharing, with load balanced front end servers and edge servers directing traffic between internal and external networks.
This document provides a comprehensive list of port requirements for various SolarWinds products. It includes the port numbers, descriptions of their uses, and in some cases which components or products they are used with. The ports vary between products and some are configurable. Refer to product documentation for specific port information.
The document discusses Microsoft Lync Server 2010 Edge architecture and scenarios. It provides an overview of the Edge topology and components, including the Access Edge Server, Web Conferencing Edge Server, and A/V Edge Server. It then summarizes scenarios for external user access, public IM federation, and NAT traversal using STUN, TURN, and ICE to relay media packets between internal and external clients.
The document discusses the process and code for building a client socket application in Java. It covers establishing a connection to a server, obtaining socket information, sending and receiving data, and closing the connection. An example client code is provided that connects to a sample server, receives data as bytes, and prints the response. The client creates a socket, gets input/output streams, reads server data into a byte array, and closes the connection when finished.
1. The document provides instructions for setting up and configuring VoIP MOS testing using Skype. It describes installing and configuring Skype on the test PC and setting up an auto-response system.
2. Steps are outlined for creating a new VoIP test script in the XCAL software, designating call parameters like connection information and timers.
3. The document also describes scheduling a test script to run and addresses required Skype security settings to allow automated calling via the XCAL software.
Microsoft lync server 2010 protocol workloads posterbigwalker
This document provides a summary of the Microsoft Lync network architecture and protocols. It includes diagrams showing the typical placement of internal and external firewalls, edge servers, directors, pools, and additional workload-specific servers. The document also lists common ports, protocols, and DNS records used across the Lync infrastructure.
SIP is an application layer protocol for initiating, modifying, and terminating multimedia sessions over IP networks. It uses URIs to address endpoints and can be used for voice, video, instant messaging and other applications. SIP includes components like user agents, servers, and gateways. Common SIP messages are INVITE to create a session, ACK, BYE, and REGISTER. SDP is used to describe multimedia sessions and carry information like media formats and ports. Common codecs are G.711 for voice and H.264 for video.
SPARSH VP248 is a high-definition VoIP phone built with superior acoustics and elegant design to provide unsurpassed audio quality and rich user experience.
Based on open-standard SIP protocol, SPARSH VP248 is interoperable with any standard SIP infrastructure such as IP-PBX, SIP Proxies, Softswitches and Stand-alone applications.
SPARSH VP248 is designed for power users, knowledge workers and managers for quick access totheadvance system features and functions. A feature-packed IP phoneenables user to work efficiently with advance call handling capabilities
Office Comunnications Server 2007 R2 PosterPaulo Freitas
This document provides an overview of the workload architecture for a unified communications solution. It includes a legend defining the various hardware components, diagrams showing traffic flows between internal and external systems, and details on DNS records, certificates, and firewall configurations required. The solution comprises multiple workloads including IM and presence, audio/video conferencing, enterprise voice, and application sharing, with load balanced front end servers and edge servers directing traffic between internal and external networks.
This document provides a comprehensive list of port requirements for various SolarWinds products. It includes the port numbers, descriptions of their uses, and in some cases which components or products they are used with. The ports vary between products and some are configurable. Refer to product documentation for specific port information.
The document discusses Microsoft Lync Server 2010 Edge architecture and scenarios. It provides an overview of the Edge topology and components, including the Access Edge Server, Web Conferencing Edge Server, and A/V Edge Server. It then summarizes scenarios for external user access, public IM federation, and NAT traversal using STUN, TURN, and ICE to relay media packets between internal and external clients.
The document discusses the process and code for building a client socket application in Java. It covers establishing a connection to a server, obtaining socket information, sending and receiving data, and closing the connection. An example client code is provided that connects to a sample server, receives data as bytes, and prints the response. The client creates a socket, gets input/output streams, reads server data into a byte array, and closes the connection when finished.
1. The document provides instructions for setting up and configuring VoIP MOS testing using Skype. It describes installing and configuring Skype on the test PC and setting up an auto-response system.
2. Steps are outlined for creating a new VoIP test script in the XCAL software, designating call parameters like connection information and timers.
3. The document also describes scheduling a test script to run and addresses required Skype security settings to allow automated calling via the XCAL software.
Programmable command-line interface API for managing operation of a network d...Tal Lavian Ph.D.
A method of managing a network device, includes providing a command-line interface application programming interface (CLI-API) compatible with a command-line interface (CLI) of the network device, receiving instructions from an application that calls one or more routines in the CLI application programming interface, and generating at least one command in response to receiving instructions from the application wherein the at least one command is compatible with the CLI of the network device. An apparatus includes a remote serial command-line interface (RS-CLI) device having a storage device capable of storing instructions, a network port capable of being connected to the network and capable of processing a network protocol stack in addition to receiving the instructions, a serial port capable of processing a serial protocol and capable of being connected to the non-application enabled network device, and a processor capable of processing instructions stored in the storage area of the RS-CLI device.
https://www.google.com/patents/US7039724?dq=US+7039724&hl=en&sa=X&ei=83dSVOCGGYvz8gXSrIHoDA&ved=0CB8Q6AEwAA
1) This document provides instructions for setting up and running a video on demand (VOD) test using the XCAL auto call feature.
2) It describes installing a video streaming server and player, adding a test video file to the server, and configuring the VOD script in XCAL to designate the video content, quality test settings, and call timers.
3) The VOD test results will display real-time transport protocol information and allow comparing the downloaded video to a reference file.
The document provides configuration steps for Verint Ultra Suite to successfully interoperate with Avaya Communication Manager 2.1. It describes testing the Ultra call recording solution, which uses Computer Telephony Integration to extract call details from Avaya Communication Manager. The solution supports active station-side recording via E1 trunks and Avaya Communication Manager API, as well as passive tapping of analog stations, E1 trunks, and VoIP. [END SUMMARY]
Most research and publications talk about layer 2 issues when it comes to VoIP. Over here we talk about VoIP security flaws that can be exploited without having physical access to the target network, i.e. attacks that can be, and are being launched through the Internet.
The document describes the protocols used by Ulteo Open Virtual Desktop, including:
- HTTP and SSL for session establishment and management between clients and servers
- RDP for displaying remote desktops and applications on clients
- HTTP for communication between servers like the Session Manager and Application Servers
It provides details on establishing client sessions, the various web services used for session management, and the JavaScript API for launching sessions from web applications.
The document describes the sequence of events that occur during IMS registration and session establishment in 3G/4G networks.
1. A UE first attaches to the GPRS network and establishes a PDP context to get an IP address. It then performs CSCF discovery and IMS registration by exchanging SIP REGISTER messages with the P-CSCF, I-CSCF and S-CSCF.
2. To establish a session, the UE sends a SIP INVITE message which is routed through CSCFs and may involve application servers based on the user's profile. The message is sent to the destination UE to initiate session negotiation.
3. Sessions involving the PST
This document provides instructions for setting up an HTTP test using the XCAL AutoCall feature. It describes how to designate the HTTP connection information and call timers, including setting the URL, repeat count, idle time, setup time, traffic time, and call count. It also explains that the throughput and summary statistics will be displayed once the test is completed.
This document provides an overview of the Session Initiation Protocol (SIP). It discusses SIP's architecture and components, including user agents, proxy servers, registrar servers, and location servers. It also explains how SIP establishes calls between two user agents on the same proxy server and how record routing affects calls between user agents on different proxy servers. Additionally, it covers related protocols like SDP for multimedia session description and ENUM for mapping phone numbers to Internet addresses.
This document provides an installation and configuration guide for TekIVR version 2.4. TekIVR is an interactive voice response (IVR) system built on SIP that allows users to create custom IVR scenarios through a graphical interface. The guide outlines system requirements, installation instructions, and how to configure settings like SIP accounts, extensions, prompts, and IVR scenarios through the TekIVR manager interface. It also provides examples of typical IVR workflows and call flows that can be created with TekIVR.
This document provides instructions for setting up and running ping and trace route tests using XCAL series AutoCall software. It describes how to designate ping connection information and call timers, including setting the destination, timeout, TTL, packet length, and count. It also explains how to set call timers for idle time, setup time, and traffic time. Finally, it notes that ping and trace route statistics can be viewed during and after the testing process to analyze results.
The document provides release notes for the VCX 9.8.15 software release. Key features of this release include Microsoft OCS and Exchange integration, simultaneous ringing, HP ProLiant DL360 G7 support, scheduled backups, ACD license enhancements, and voicemail return receipt. The release also includes various enhancements to IPM, IMC, extensions, replication setup, and the global directory.
This document provides instructions for setting up automated test calls (AutoCalls) in XCAL drive test software to send and receive multimedia messages (MMS). It describes how to create an AutoCall scenario to send MMS, including configuring timer settings and MMS message details. It also covers creating a scenario to receive MMS. The document includes a table of common MMS server settings for different mobile operators.
This document describes how to configure a zone-based policy firewall (ZPF) on router R3 to allow internal hosts to access external resources while blocking external hosts from accessing internal resources. It involves verifying basic network connectivity, creating firewall zones, defining traffic classes and access lists, specifying firewall policies, applying policies to zone pairs, and testing functionality from the internal and external zones.
PLNOG 9: Emil Gągała - Fast Service Restoration PROIDEA
This document discusses providing fast service restoration for 2547 VPN services. It focuses on using local protection to provide sub-50 millisecond connectivity recovery when the egress PE router fails. The key challenges are that the point of local repair (PLR) router does not have service label mappings and the backup node needs to correctly interpret service labels. The solution details how to distribute service label mappings to address these problems.
This document discusses network programming in Python using sockets. It explains that sockets allow communication across networks and the Python socket module provides an interface to work with sockets. It then describes how to create server and client sockets, including binding, listening, accepting connections, and sending/receiving data. It also covers different socket address families like AF_INET for IPv4 and provides code examples for a basic echo server and handling multiple clients using threads or processes.
The document describes a new type of foundation called a MeshWORKS foundation. [1] MeshWORKS foundations focus on aligning and integrating diverse resources to achieve common goals and "third win" outcomes. [2] They prioritize enabling and empowering all contributing elements rather than promoting a single image or cause. [3] The defining purpose of MeshWORKS foundations is to search for objectives that can mesh people and organizations together for positive impact.
Hello Mr. Douglas, this is ABC calling with an important message about your recent purchase. You may recall purchasing a new laptop computer from us 2 weeks ago. I'm pleased to inform you that we are now offering an optional 14-day extended warranty on all laptop purchases for only $49. This extended warranty will double your original 1 year warranty and provide you with complete accident protection as well as on-site service if needed. Would you like me to add the 14 day extended warranty to your original purchase? Just say yes and I can process that for you right now. Say no if you're not interested.
Customer: No thanks, I'm good with the original warranty.
El documento presenta los resultados académicos de 10 estudiantes de un centro educativo en diferentes asignaturas como matemáticas, español y ciencias, así como su promedio general. También incluye tablas sobre los deportes favoritos de un grado y los valores con los que los estudiantes se identifican en mayor medida.
The document summarizes Juniper Networks' presentation at a press and analyst conference on May 17, 2010. The presentation focused on introducing Juniper's vision for "the new network for the data center", which is simplified, automated, and secured. Specific announcements included new switches, routers, automation applications built on Juniper's Junos Space network management platform, as well as partnerships with IBM, Dell, and VMware. Real-world customer examples were presented to demonstrate the performance, efficiency, and cost benefits of Juniper's approach compared to competitors like Cisco.
Open Advocacy - Citizens Connected Goes Open SourceMyplanet Digital
The document invites citizens to a code sprint on December 7-8 in Toronto to work on open sourcing the Citizens Connected project and its Drupal modules for civic engagement and advocacy. It outlines the agenda which includes introducing Citizens Connected (CitCon), discussing its future as an open source project, comparing its Drupal modules, highlighting key features, and what remains to be done. Attendees are encouraged to come to the code sprint to contribute.
The future of work is uncertain but will likely involve more transparency, flat and virtual structures, competitiveness, and on-demand work. Jobs and industries will continue to change rapidly as new technologies emerge. Success will depend on individuals distinguishing themselves through skills, innovation, and the ability to work virtually across borders in competitive global marketplaces. The future workplace may involve less traditional careers and more independent contracting and virtual team-based project work.
This document reports on Remote Procedure Call (RPC) and distributed systems. It provides background on RPC, describing it as a technique that allows a program to execute a subroutine in another address space, such as on another computer, without explicitly coding message passing details. It then gives timelines and information flows for how RPC works. The document also discusses socket programming as an implementation of RPC, showing code examples of a socket server and client that demonstrate how sockets allow message passing between processes similarly to RPC.
Programmable command-line interface API for managing operation of a network d...Tal Lavian Ph.D.
A method of managing a network device, includes providing a command-line interface application programming interface (CLI-API) compatible with a command-line interface (CLI) of the network device, receiving instructions from an application that calls one or more routines in the CLI application programming interface, and generating at least one command in response to receiving instructions from the application wherein the at least one command is compatible with the CLI of the network device. An apparatus includes a remote serial command-line interface (RS-CLI) device having a storage device capable of storing instructions, a network port capable of being connected to the network and capable of processing a network protocol stack in addition to receiving the instructions, a serial port capable of processing a serial protocol and capable of being connected to the non-application enabled network device, and a processor capable of processing instructions stored in the storage area of the RS-CLI device.
https://www.google.com/patents/US7039724?dq=US+7039724&hl=en&sa=X&ei=83dSVOCGGYvz8gXSrIHoDA&ved=0CB8Q6AEwAA
1) This document provides instructions for setting up and running a video on demand (VOD) test using the XCAL auto call feature.
2) It describes installing a video streaming server and player, adding a test video file to the server, and configuring the VOD script in XCAL to designate the video content, quality test settings, and call timers.
3) The VOD test results will display real-time transport protocol information and allow comparing the downloaded video to a reference file.
The document provides configuration steps for Verint Ultra Suite to successfully interoperate with Avaya Communication Manager 2.1. It describes testing the Ultra call recording solution, which uses Computer Telephony Integration to extract call details from Avaya Communication Manager. The solution supports active station-side recording via E1 trunks and Avaya Communication Manager API, as well as passive tapping of analog stations, E1 trunks, and VoIP. [END SUMMARY]
Most research and publications talk about layer 2 issues when it comes to VoIP. Over here we talk about VoIP security flaws that can be exploited without having physical access to the target network, i.e. attacks that can be, and are being launched through the Internet.
The document describes the protocols used by Ulteo Open Virtual Desktop, including:
- HTTP and SSL for session establishment and management between clients and servers
- RDP for displaying remote desktops and applications on clients
- HTTP for communication between servers like the Session Manager and Application Servers
It provides details on establishing client sessions, the various web services used for session management, and the JavaScript API for launching sessions from web applications.
The document describes the sequence of events that occur during IMS registration and session establishment in 3G/4G networks.
1. A UE first attaches to the GPRS network and establishes a PDP context to get an IP address. It then performs CSCF discovery and IMS registration by exchanging SIP REGISTER messages with the P-CSCF, I-CSCF and S-CSCF.
2. To establish a session, the UE sends a SIP INVITE message which is routed through CSCFs and may involve application servers based on the user's profile. The message is sent to the destination UE to initiate session negotiation.
3. Sessions involving the PST
This document provides instructions for setting up an HTTP test using the XCAL AutoCall feature. It describes how to designate the HTTP connection information and call timers, including setting the URL, repeat count, idle time, setup time, traffic time, and call count. It also explains that the throughput and summary statistics will be displayed once the test is completed.
This document provides an overview of the Session Initiation Protocol (SIP). It discusses SIP's architecture and components, including user agents, proxy servers, registrar servers, and location servers. It also explains how SIP establishes calls between two user agents on the same proxy server and how record routing affects calls between user agents on different proxy servers. Additionally, it covers related protocols like SDP for multimedia session description and ENUM for mapping phone numbers to Internet addresses.
This document provides an installation and configuration guide for TekIVR version 2.4. TekIVR is an interactive voice response (IVR) system built on SIP that allows users to create custom IVR scenarios through a graphical interface. The guide outlines system requirements, installation instructions, and how to configure settings like SIP accounts, extensions, prompts, and IVR scenarios through the TekIVR manager interface. It also provides examples of typical IVR workflows and call flows that can be created with TekIVR.
This document provides instructions for setting up and running ping and trace route tests using XCAL series AutoCall software. It describes how to designate ping connection information and call timers, including setting the destination, timeout, TTL, packet length, and count. It also explains how to set call timers for idle time, setup time, and traffic time. Finally, it notes that ping and trace route statistics can be viewed during and after the testing process to analyze results.
The document provides release notes for the VCX 9.8.15 software release. Key features of this release include Microsoft OCS and Exchange integration, simultaneous ringing, HP ProLiant DL360 G7 support, scheduled backups, ACD license enhancements, and voicemail return receipt. The release also includes various enhancements to IPM, IMC, extensions, replication setup, and the global directory.
This document provides instructions for setting up automated test calls (AutoCalls) in XCAL drive test software to send and receive multimedia messages (MMS). It describes how to create an AutoCall scenario to send MMS, including configuring timer settings and MMS message details. It also covers creating a scenario to receive MMS. The document includes a table of common MMS server settings for different mobile operators.
This document describes how to configure a zone-based policy firewall (ZPF) on router R3 to allow internal hosts to access external resources while blocking external hosts from accessing internal resources. It involves verifying basic network connectivity, creating firewall zones, defining traffic classes and access lists, specifying firewall policies, applying policies to zone pairs, and testing functionality from the internal and external zones.
PLNOG 9: Emil Gągała - Fast Service Restoration PROIDEA
This document discusses providing fast service restoration for 2547 VPN services. It focuses on using local protection to provide sub-50 millisecond connectivity recovery when the egress PE router fails. The key challenges are that the point of local repair (PLR) router does not have service label mappings and the backup node needs to correctly interpret service labels. The solution details how to distribute service label mappings to address these problems.
This document discusses network programming in Python using sockets. It explains that sockets allow communication across networks and the Python socket module provides an interface to work with sockets. It then describes how to create server and client sockets, including binding, listening, accepting connections, and sending/receiving data. It also covers different socket address families like AF_INET for IPv4 and provides code examples for a basic echo server and handling multiple clients using threads or processes.
The document describes a new type of foundation called a MeshWORKS foundation. [1] MeshWORKS foundations focus on aligning and integrating diverse resources to achieve common goals and "third win" outcomes. [2] They prioritize enabling and empowering all contributing elements rather than promoting a single image or cause. [3] The defining purpose of MeshWORKS foundations is to search for objectives that can mesh people and organizations together for positive impact.
Hello Mr. Douglas, this is ABC calling with an important message about your recent purchase. You may recall purchasing a new laptop computer from us 2 weeks ago. I'm pleased to inform you that we are now offering an optional 14-day extended warranty on all laptop purchases for only $49. This extended warranty will double your original 1 year warranty and provide you with complete accident protection as well as on-site service if needed. Would you like me to add the 14 day extended warranty to your original purchase? Just say yes and I can process that for you right now. Say no if you're not interested.
Customer: No thanks, I'm good with the original warranty.
El documento presenta los resultados académicos de 10 estudiantes de un centro educativo en diferentes asignaturas como matemáticas, español y ciencias, así como su promedio general. También incluye tablas sobre los deportes favoritos de un grado y los valores con los que los estudiantes se identifican en mayor medida.
The document summarizes Juniper Networks' presentation at a press and analyst conference on May 17, 2010. The presentation focused on introducing Juniper's vision for "the new network for the data center", which is simplified, automated, and secured. Specific announcements included new switches, routers, automation applications built on Juniper's Junos Space network management platform, as well as partnerships with IBM, Dell, and VMware. Real-world customer examples were presented to demonstrate the performance, efficiency, and cost benefits of Juniper's approach compared to competitors like Cisco.
Open Advocacy - Citizens Connected Goes Open SourceMyplanet Digital
The document invites citizens to a code sprint on December 7-8 in Toronto to work on open sourcing the Citizens Connected project and its Drupal modules for civic engagement and advocacy. It outlines the agenda which includes introducing Citizens Connected (CitCon), discussing its future as an open source project, comparing its Drupal modules, highlighting key features, and what remains to be done. Attendees are encouraged to come to the code sprint to contribute.
The future of work is uncertain but will likely involve more transparency, flat and virtual structures, competitiveness, and on-demand work. Jobs and industries will continue to change rapidly as new technologies emerge. Success will depend on individuals distinguishing themselves through skills, innovation, and the ability to work virtually across borders in competitive global marketplaces. The future workplace may involve less traditional careers and more independent contracting and virtual team-based project work.
This document reports on Remote Procedure Call (RPC) and distributed systems. It provides background on RPC, describing it as a technique that allows a program to execute a subroutine in another address space, such as on another computer, without explicitly coding message passing details. It then gives timelines and information flows for how RPC works. The document also discusses socket programming as an implementation of RPC, showing code examples of a socket server and client that demonstrate how sockets allow message passing between processes similarly to RPC.
1. The document describes IMS call flows for registration, session establishment, and interworking between IMS and CS networks.
2. Key steps in registration include bearer level registration with the GSN, PDP context activation, CSCF discovery, and application level registration with the S-CSCF which involves authentication.
3. Session establishment involves SIP INVITE messages being routed through CSCFs and optionally application servers based on filter criteria before reaching the terminating user.
Session Initiation Protocol (SIP) is an application layer protocol for setting up and managing multimedia communication sessions over IP networks. It allows users to initiate, modify and terminate multimedia sessions that include voice, video and messaging applications. SIP supports mobility through proxy servers that can forward calls to a user's current location. Common security threats to SIP include registration hijacking, message modification and denial of service attacks. Recommended security mechanisms include TLS for hop-by-hop security, S/MIME for end-to-end encryption, and digest authentication.
This document provides examples of how to obtain MAC and IP address accounting information from Cisco routers using SNMP. MAC address accounting tracks packet and byte counts for each unique MAC address on a LAN interface. IP address accounting tracks packet and byte counts for source and destination IP address pairs. The document outlines how to retrieve this information using SNMP commands to access the CISCO-IP-STAT-MIB and OLD-CISCO-IP-MIB, including using checkpointing to copy the active IP accounting database to a stable checkpoint database.
This document provides troubleshooting best practices for Genesys Voice Platform (GVP) 8.x. It discusses how GVP applications communicate with each other using SIP, describes common SIP error codes and how to track calls using identifiers in Genesys logs. The document explains how to enable custom logging, save temp files, and manually produce dump and core files. It also reviews using Wireshark to analyze network traffic and addresses common questions and problems.
Design an Implementation of A Messaging and Resource Sharing Softwarenilabarai
In this article it has been looked how to program using sockets by implementing an echo server along with a client that is used to send and receive string messages. It will start off by giving a quick introduction to TCP/IP fundamentals and then explain how sockets fit into the diagram. Most network application can be divided into two pieces: a client and a server. A client is the side that initiates the communication process, where as the server responds to incoming client requests. There would be contains three types of service from this software, that is Message Transfer service, Voice Transfer service and File Transfer service. All this sorts are the design issue of this software.
From MSS to TelScale - Mobicents Summit 2011telestax
The document discusses Mobicents Sip Servlets Stack (MSS) achievements and introduces TelScale as the next generation stack. Key points include:
- MSS 2.X added Servlet 3.0 support, started a CDI Telco framework and testing with Arquillian.
- TelScale will provide container abstraction, HA independence, and focus on cloud deployment and operations.
- Future roadmaps include cloud communication features in 2.X and SIP Servlets 2.0 work in 3.X.
This document provides an overview of the transport layer and various transport layer protocols. It discusses socket programming and describes TCP and UDP in detail. The key points covered are:
- An introduction to the transport layer and its role in multiplexing communication between processes.
- Socket programming interfaces that allow communication between applications and transport protocols like TCP/IP.
- TCP provides reliable, connection-oriented data transfer with flow control and congestion control. UDP provides simpler connectionless datagram delivery.
- Details on TCP headers, connection establishment and closing, transmission policies, and how it provides reliable data transfer over unreliable IP.
CTU June 2011 - Exchange and Lync IM IntegrationSpiffy
This document provides an agenda and overview for a presentation on integrating Exchange and Lync instant messaging (IM) through Outlook Web App (OWA). The presentation will include screenshots demonstrating the integrated interface, requirements for Exchange 2010 SP1 and Lync 2010, explanations of the necessary components, and a demo with Q&A. It outlines installing and configuring the Web Service Provider on the Exchange Client Access Server and adding the Exchange server as a trusted application in Lync to enable the integration.
Upon reading the document, the key steps in a router's start-up process can be summarized as follows:
1. When power is applied, the router performs a power-on self-test and loads the bootstrap code from ROM to initialize hardware and find the IOS image.
2. The IOS image is then loaded from flash memory or another source such as TFTP into RAM where it is decompressed and executed.
3. The startup configuration is loaded, typically from NVRAM. If no configuration is present, the router enters setup mode to configure initial settings.
This document defines the components and methods used in SIP including user agents, servers, methods, SDP fields, and call flows. It describes UAs, proxies, registrars, redirect and location servers. It outlines the REGISTER, INVITE, ACK, CANCEL, OPTIONS, and BYE methods and how Cisco gateways can send and receive them. It also summarizes SDP fields, DTMF relay options, error codes, dialpeer configuration, and SIP UA commands.
Remote Procedure Call (RPC) is a protocol that one program can use to request a service from a program located in another computer in a network without having to understand network details. (A procedure call is also sometimes known as a function call or a subroutine call.) RPC uses the client/server model. The requesting program is a client and the service-providing program is the server. Like a regular or local procedure call, an RPC is a synchronous operation requiring the requesting program to be suspended until the results of the remote procedure are returned. However, the use of lightweight processes or threads that share the same address space allows multiple RPCs to be performed concurrently.
This document provides an overview of the transport layer and protocols TCP and UDP. It discusses how the transport layer provides communication between processes on different hosts using port numbers for multiplexing. TCP provides reliable, connection-oriented transport using mechanisms like flow control, congestion control, and reliable data transfer with sequence numbers and acknowledgments. UDP provides a simpler, connectionless datagram service without reliability. The document outlines TCP and UDP header formats and explains TCP connection establishment and closing procedures. It also describes TCP congestion control algorithms like slow start, congestion avoidance, fast retransmit, and timeouts for loss recovery.
1. The document introduces Session Initiation Protocol (SIP), explaining that it is an application layer signaling protocol for initiating, modifying, and terminating multimedia communication sessions over IP such as voice and video calls.
2. It describes why SIP is used, including for conferencing, distance learning, video conferencing, instant messaging, and voice calls. It also outlines the main components of a SIP network including user agents, proxies, and redirects servers.
3. The document provides an overview of how SIP works by outlining the signaling process for registration, call setup and teardown, redirection, and media routing between user agents.
This lab guide provides instructions for completing several labs that demonstrate an Intelligent WAN (IWAN) solution. The labs utilize a virtual lab environment containing routers, servers, and PCs in a data center and branch office. Students will navigate the lab topology, generate application traffic, and configure Cisco Prime Infrastructure and other components. The objective is to understand the IWAN architecture and how it optimizes application performance over the WAN.
The document describes several registration and de-registration flows for IP Multimedia Subsystem (IMS). The key steps are:
1. For registration, the UE requests P-CSCF information from DHCP and DNS servers, then sends a register request to the P-CSCF which assigns a S-CSCF and retrieves the user profile from HSS.
2. Periodic re-registration follows the same process to refresh the registration.
3. For de-registration, the UE or network send a register request with expiration time of zero, removing the registration.
The document provides an overview of the Microservice Bus (MSB) which provides a communication infrastructure to support microservice architecture. Key components of MSB include service registration using a registry like Consul, client-side and server-side service discovery, and API gateways. MSB also includes tools like the Kube2MSB registrator to automatically register services deployed on Kubernetes. Examples are provided for running MSB without and within the ONAP Operations Manager (OOM). The future vision is presented of integrating service mesh technologies into the ONAP Microservice Architecture (OMSA) for carrier-grade microservices.
This document discusses managed device deployment at branch offices using Aruba branch controllers. It provides an overview of how branch controllers connect to a master controller via an internet modem and establish communication. It also covers branch controller and VPN concentrator configuration in Aruba OS versions 6.x and 8.x, including initial setup, zero touch provisioning, and debugging tools. Additional topics include address pool management for VLANs, tunnels, NAT, and DHCP to allow for dynamic IP assignment at branch office deployments.
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Converged office engineering detail
1. Converged Office Engineering Session
Converged Office Client Startup
As initially demonstrated in the live demo in the lab with the configuration below, in the first
instance, when the OCS client has logged in, the OCS Front End Server informs the CS1000 that
the user login has taken place. The CS1000 has to be informed of this in order to be able to route
calls to this user. It is the SIP CTI Front End Signalling Server part of the CS1000 that is
informed of this via a SIP INVITE message that comes from the Application Proxy Server. The
process is as follows:
NRS/SPS Active
Directory
Call Server
3 OCS2007
4 FE Server
2
6 5 7
7 1
SIP CTI Front End 7
Signalling Server
MCM/Application Proxy
Figure 5.4
Figure 5.4 shows the MOC (Microsoft Office Communicator) client startup and process of
registration and subsequent control request of the telephone for converged desktop functionality.
1. The Office Communicator client starts and sends a SIP INVITE message to the OCS FE
Server asking for control of the telephone.
2. The OCS FE Server validates the request (via AD) and forwards it to the OCS
Application Proxy Server.
3. The OCS Application Proxy Server the forwards the INVITE to the NRS/SPS
4. The NRS responds with the “Moved Temporarily” as it looks up, and then provides the
IP address of the appropriate Node (SIP CTI Server)
2. 5. The OCS Application Proxy Server can now send the INVITE to the Signalling Server
Node IP
6. The Signalling Server then processes the request with the Call Server for the phone DN
acquisition.
7. The Signalling Server sends back a “200 OK” message to the OCS Application Server,
forwards back to the OCS FE Server and then to the OCS Client PC. The phone is now
controlled.
Traffic monitoring of this telephone control request process was initiated on initial login, and
here we see the client startup and registration process in more detail.
Instance-Id: 00000 A66
Client PC IP address
Direction: incoming
Peer: 47.164.154.177:1219 User SIP URI
Message-Type: request
Start-Line : INVITE sip:4115@demolab .com SIP/2.0
From: <sip:usera@demolab.com>;tag=b0434 a26b8;epid =f60275d516
To: <sip:4115@demolab.com>
CSeq: 1 INVITE
…
Message-Body : <?xml version="1.0"?>
<RequestSystemStatus xmlns ="http://www.ecma-international .org/standards /ecma-
323/csta/ed3"><extensions ><privateData><private><lcs:line
xmlns:lcs="http://schemas.microsoft.com/Lcs/2005/04/RCCExtension ">tel:+33164554115 ;
ext=4115</lcs:line></
Figure 5.5
1. SIP INVITE From The Client PC
As indicated in Figure 5.5, a trace on the client PC, shows that RCC mode has only one
INVITE message when starting the Office Communicator. Each call is associated to
INFO messages and not INVITE messages. In this case, there will only be one session
during the OC call duration. Additionally, the OC sends polling (INVITE) messages
every 10 minutes to ensure the telephone is still controlled.
3. Instance-Id: 00000 A6E
Direction: outgoing Application Proxy Server
Peer: 47.164.113.136:5060
Message-Type: request
Start-Line : INVITE sip:4115@demolab .com SIP/2.0
From: “User A"<sip:usera@demolab .com>;tag=b0434a26b8;epid=f60275 d516
To: <sip:4115 @demolab .com>
CSeq: 1 INVITE
Call-ID: b094883980 e94e0490ed9174a99f481d
ms-user-data: ms-publiccloud =true;ms-federation =true
Record-Route: <sip:DemoOcsFeSrv1.demolab .com;transport=tcp;ms-role-rsto;
lr>;tag=B62E609FD01E60435 DE8508 CC53B6483
Via: SIP/2.0/TCP
47.164.113.137:2608;branch=z9hG4bK932C8642.99918 F3C;branched =FALSE
Max-Forwards: 69
Figure 5.6
2. Front End Server forwards the INVITE to the Application Proxy Server
Message monitoring on the FE server on Figure 5.6 shows that the INVITE is then
forwarded onto the Application Proxy Server. It is important to note here that the routing
entry for the Proxy Server has to be defined in the FE Server.
Instance-Id: 000000 FE
Direction: outgoing
Peer: 47.164.113.108:5060 NRS IP Address
Message-Type: request
Start-Line : INVITE
sip:4115@demolab .com:5060;maddr =47.164.113.108;transport=tcp;x-nt-netfeature =
x-nt-redirect SIP/2.0
From: “User A"<sip:usera@demolab .com>;tag=b0434a26b8;epid=f60275 d516
To: <sip:4115 @demolab .com>
CSeq: 1 INVITE
Figure 5.7
3. Application Proxy Forwards to the NRS
As shown in figure 5.7, the INVITE is then forwarded on to the NRS for correct routing
within the telephony network. The NRS IP address is derived from the configuration
parameter in the MCM.
4. Instance-Id: 000000 FF
Direction: incoming A
Peer: 47.164.113.108:5060
Message-Type: response
Start-Line : SIP/2.0 302 Moved Temporarily
From: “User A"<sip:usera@demolab .com>;tag=b0434a26b8;epid=f60275 d516
To: <sip:4115@demolab .com>;tag=16951
CSeq: 1 INVITE B
…
Contact: <sip:4115;phonecontext =
cdp.udp@ctflab.net:5060 ;maddr=47.164.113.106;transport =tcp;xnt-
net-feature =x-nt-redirect;x-nt-redirect=redirect-server>
Content-Length : 0
Message-Body: –
$$end_record
Figure 5.8
4. NRS directs to the appropriate SIP CTI Server
Monitoring on the Application Proxy Server as depicted in Figure 5.8 shows that the
NRS processes the INVITE message and sends it back to the Proxy Server with the
appropriate Node IP address (SIP CTI Server) responsible for handling the DN range
(41xx) (A). The Proxy Server then needs to contact this SIP CTI Server and re-issue the
same INVITE back to this server.
Instance-Id: 00000101
Direction: outgoing
Peer: 47.164.113.106:5060 Node IP address of SIP CTI
Message-Type: request Front End Signalling Server
Start-Line: INVITE sip :4115;phone-
context=cdp.udp@demolab .comt:5060;maddr=47.164.113.106;transport=tcp;x-ntnet-
feature=x-nt-redirect;x-nt-redirect=redirect-server SIP /2.0
From: “User A"<sip:usera@demolab .com>;tag=b0434a26b8;epid=f60275d516
To: <sip:4115@demolab.com>
CSeq: 1 INVITE
1
Call-ID: b094883980 e94e0490ed9174a99f481d 2
Record-Route:
<sip:DemolabOcsProxySrv .demolab .com;transport=tcp;lr>;tag=8C65C54CB3D548F691E917450 BD
A37AF
Via: SIP/2.0/TCP 47.164.113.136:1613;branch=z9hG4bK3560E034.F1D7AAD1;branched =TRUE
Max-Forwards : 68
Content-Length: 321
Via: SIP/2.0/TCP 47.164.113.137:2608;branch=z9hG4bK932C8642.99918F3C;branched=FALSE;ms-
receivedport=
2608;ms-received-cid=700
Via: SIP/2.0/TCP 47.164.154.177:1219;ms-received-port=1219;ms-received -cid=4C00
ms-user-data: ms-publiccloud =true;ms-federation=true 3
Record-Route: <sip:DemolabOcsFeSrv 1.demolab .com;transport=tcp;ms-role-rsto;
lr>;tag=B62E609FD01E60435DE8508CC53B6483
Contact: <sip:usera@demolab.com;opaque=user:epid:kI5Vslm4gl2F2-3nWKKnhQAA ;gruu>
User-Agent: UCCP/2.0.6362.36 OC/2.0.6362.36 (Microsoft Office Communicator )
Content-Disposition : signal;handling=required
5. Figure 5.9
5. Application Proxy Server sends the INVITE back to the appropriate SIP CTI
Server
The trace on Figure 5.9 shows points of note:
1. The Signalling Server constructs an EPID (Endpoint Identifier) to store information
related to the DN (Dialled Number). It associates the SIP URI of the user with this epid.
2. The Signalling Server now is aware of the Recorded Route of the Application Proxy
Server in the OCS environment.
3. The Signalling Server also is now aware of Recorded Route of the OCS Front End Server
This information will be necessary to build the SIP INFO message related to an RCC call for this
DN. (In this case 4115). Also, the SIP INFO will contain everything required in order for it to
route the the correct user client PC and therefore the MCM will not have to perform the Caller
DN number lookup operation.
A
ELAN32 I MTYP=3B IACR TN=0 TIME=12:35:08
ELAN32 IN 53C56D7F OUT 53C56D80 QSIZE 00000000
ELAN32 03 20 00 00 00 00 1 E 3B 00 A6 00 00 95 01 05 36 02 41 15 E6
ELAN32 0C BF EE 01 FF FF FF 00 00 0 F FF 00 00
ELAN32 O MTYP=3C IACS TN=0 TIME=12:35:08
ELAN32 IN 53C56D81 OUT 00000000 QSIZE 00000000
ELAN32 03 27 00 00 00 00 1 E 3C 00 A6 00 00 95 01 05 36 02 41 15 E6
ELAN32 0C BF EE 01 FF FF FF 00 00 0 F FF 00 00 37 02 61 00 AA 01 00
B
Figure 5.10
6. The Signalling Server processes the INVITE message
Figure 5.10 indicates via internal CS1000 switch messaging the processed INVITE data
being passed to the Call Server. There are some points worth noting:
A. the DN number (in this case 4115) has to match the number that is configured for
the user in Active Directory.
B. When the IACS (Infosec Assurance and Certification Services) contains the Status
Info Element “AA 01 00” string this is the indication that the Signalling Server
successfully acquired the telephone set associated with DN 4115. Any other string
means that the acquisition was not successful.
6. Instance-Id: 00000104 CS1000 Node IP address
Direction: incoming
Peer: 47.164.113.106:5060
Message-Type: response
Start-Line : SIP/2.0 200 OK
From: “User A"<sip:usera@demolab .com>;tag=b0434a26b8;epid=f60275 d516
To: <sip:4115 @ctflab.net>;tag=19042 ee0-6a71a42f-13c4-40030 -1f8cde-706ef19f-1f8cde
CSeq: 1 INVITE
Figure 5.11
7. The Signalling Server Responds to the INVITE
When the “200 OK” message is received from the Signalling Server, the Application
Proxy Server sends it to the FE Server, which then forwards it on to the Client PC
running the Office Communicator.
So this exercise indicates how the CS1000 and OCS environments have integrated and initially
communicated by informing each other of users details, and mapping SIP URI’s to DN’s for
cohesive communication.
Converged Office call scenario in Remote Call Control
Next, as demonstrated in the lab in the second instance, with the configuration detailed below, a
user logged in as RCC mode, has access to a fully integrated scenario in which the client has
control of the phone. Here we explore how in an RCC call, (a), this initial information would be
used in facilitating this transaction.
Figure 5.12 shows the flow of an incoming RCC call.
7. Active
Directory
SIP CTI Front End
Signalling Server
1 3
OCS2007
4 FE Server
2
5
MCM Proxy 3.0
RCC User Phone
OCS Client A
(4116)
Figure 5.12
Calling number is 4116
Called number is 4115
The message flow is as follows:
1. The call is presented to the controlled telephone. The Call Server then sends this message
(via the CS1000 internal mechanism) to the Signalling Server indicating that the call is
presented to the controlled telephone.
2. The SIP CTI Front End Singalling Server then initiates the SI INFO (not INVITE) to the
OCS Application Proxy Server (on which the MCM resides)
3. The MCM then tries to lookup the Calling number.
4. The OCS Application Proxy Sever then forwards this INFO message to the OCS Front
End Server
5. The OCE FE Server then sends this INFO message to the client PC running the OCS
client.
On the previous live demo in the lab, internal message monitoring and SIP messaging on the
various elements of the Converged Office Solution, Call Server, SIP Proxy Server, OCS
Application Proxy was enabled on initiation of this RCC call, and the breakdown of this
messaging structure is analysed from Figure 5.13.
8. ELAN32 O MTYP=1A USM TN=096 0 00 16 TIME=13:04:25
A ELAN32 IN 53FAFF29 OUT 00000000 QSIZE 00000000 Internal messaging from the Call Server to
Singalling Server indicating call is presented
ELAN32 03 33 00 00 61 00 16 1 A 00 00 00 00 37 02 61 00 36 02 41 15 to the controlled telephone
ELAN32 3B 01 08 38 01 03 3A 02 61 01 39 02 41 16 3C 01 08 96 04 00
ELAN32 00 0B A5 5F 07 0B 01 14 08 0D 04 19
The Signalling Server then generates the
INFO sip:userA@demolab.com;opaque=user:epid:kI5Vslm4gl2F2-3nWKKnhQAA;gruu SIP/2.0 SIP INFO message and
Corresponding
From: <sip:4115@demolab.com>;tag=19042ee0-6a71a42f-13c4-40030-1f8cde-706ef19f-1f8cde Application Proxy
sends it to the OCS
Server (MCM)
B To: “User A"<sip:usera@demolab.com>;tag=b0434a26b8;epid=f60275d516
Call-ID: b094883980 e94e0490ed9174a99f481d
CSeq: 29 INFO
Via: SIP/2.0/UDP 47.164.113.106:5060;branch=z9hG4bK-1f93be-7b591ea8-55d0a836
Content-Disposition: signal;handling=required
Max-Forwards: 70
The OCS Application Proxy Server and Front
Supported: 100rel,x-nortel-sipvc,replaces,timer
End Server route details are derived from
User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.00.31 The EPID (end point ID )
Route: <sip:DemoLabOcsProxySrv .demolab.com;transport=tcp;lr>;tag=8C65C54CB3D548F691E917450BDA37AF
Route: <sip:DemoLabOcsFeSrv1.demolab.com;transport=tcp;lr;ms-role-rs-
to>;tag=B62E609FD01E60435DE8508CC53B6483
Contact: <sip:4115;phone -context=cdp.udp@ctflab.net:5060;maddr=47.164.113.106;transport=tcp;x-nt-net-feature=x-nt-
redirect;x-ntredirect=
redirect-server>
Content-Type: application/csta+xml
Figure 5.13
1. Incoming Call presented to the controlled telephone
A. Shows internal messaging in the CS1000. The dialed digits (4115) are passed from the
Call Server to the SIP CTI Front End Signalling Server.
B. SIP tracing on the Signalling Server out to the Application Proxy server indicates how the
stored information from the previous handshaking communication between the two
systems is now utilized.
ProcessRequestFromGateway Request = INFO
XmlModificator .Apply: invoked for DeliveredEvent
LookupUserData: search for msRTCSIPLine =
tel:*;ext = 4116
LookupUserName: trying to find in AD cache
AD Cache: getValues : indexName : "msRTCSIPLine=
tel:*;ext" indexValue : "4116"
Debug: ProcessRequestFromGateway : TR87
SIP: Sent by MCM:
Figure 5.14
2. The INFO message is received by the OCS Application Proxy Server (SIP message
sent by MCM)
As seen in Figure 5.14 In RCC mode, the INFO messages already contain the required
information to e able to route the user DN and the SIO URL. The Siganlling Server stores
9. this information inside the EPID when the telephone is acquired. MCM does not have to
lookup the called number to SIP URI for an RCC call. However, it can perform a lookup
to the calling number.
Request: INFO sip:usera@demolab.com;opaque=user:epid:kI5Vslm4gl2F2-3nWKKnhQAA;gruu
via: SIP/2.0/TCP 47.164.113.106:5060;branch=z9hG4bK-1f93be-7b591ea8-55d0a836;
ms-received-port=3597;ms-receivedcid=1000
from: <sip:4115@demolab.com>;tag=19042ee0-6a71a42f-13c4-40030-1f8cde-706ef19f-1f8cde
to: “UserA"<sip:usera@demolab.com>;tag=b0434a26b8;epid=f60275d516
Info message received from
call-id: b094883980e94e0490ed9174a99f481d Signalling Server
cseq: 29 INFO
?xml version="1.0" encoding="UTF-8"?><DeliveredEvent xmlns="http://www.ecma-international.org/standards/ecma-
323/csta/ed3"><monitorCrossRefID>166</monitorCrossRefID><connection><callID>2981</callID>
<deviceID>tel:+33164554115 ;ext=4115</deviceID></connection><alertingDevice><deviceIdentifier>tel:+33164554115 ;ext=4115</
deviceIdentifier></alertingDevice>
<callingDevice><deviceIdentifier>tel:4116;phonecontext=cdp.udp</deviceIdentifier></callingDevice><calledDevice>
<deviceIdentifier>tel:+33164554115 ;ext=4115</deviceIdentifier></calledDevice><lastRedirectionDevice><notRequired/></lastRedir
ectionDevice><localConnectionInfo>alerting</localConnectionInfo><cause>normal</cause></DeliveredEvent>
1/11/2008 3:57:03 PM: 3.0.1.77: Debug: ProcessRequestFromGateway Request = INFO
1/11/2008 3:57:03 PM: 3.0.1.77: Debug: XmlModificator.Apply: invoked for DeliveredEvent
1/11/2008 3:57:03 PM: 3.0.1.77: Debug: LookupUserData: search for msRTCSIPLine=
tel:*;ext = 4116 AD Lookup
1/11/2008 3:57:03 PM: 3.0.1.77: Debug: LookupUserName: trying to find in AD cache
1/11/2008 3:57:03 PM: 3.0.1.77: Debug: AD Cache: getValues: indexName: "msRTCSIPLine=
tel:*;ext" indexValue: "4116“
Request: INFO sip:usera@demolab.com;opaque=user:epid:kI5Vslm4gl2F2-3nWKKnhQAA;gruu
via: SIP/2.0/TCP 47.164.113.106:5060;branch=z9hG4bK-1f93be-7b591ea8-55d0a836;ms-received-port=3597;ms-received-
cid=1000
from: <sip:4115@demolab.com>;tag=19042ee0-6a71a42f-13c4-40030-1f8cde-706ef19f-1f8cde
to: “User A"<sip:usera@demolab.com>;tag=b0434a26b8;epid=f60275d516
call-id: b094883980e94e0490ed9174a99f481d Info message sent to FE Server
cseq: 29 INFO
?xml version="1.0" encoding="UTF-8"?><DeliveredEvent xmlns="http://www.ecma-international.org/standards/ecma-
323/csta/ed3"><monitorCrossRefID>166</monitorCrossRefID><connection><callID>2981</
callID><deviceID>tel:+33164554115 ;ext=4115</
deviceID></connection><alertingDevice><deviceIdentifier>tel:+33164554115 ;ext=4115</deviceIdentifier></
alertingDevice><callingDevice><
deviceIdentifier>tel:+33169554116 ;ext=4116</deviceIdentifier></ A
callingDevice><calledDevice><deviceIdentifier>tel:+33164554115 ;ext
=4115</deviceIdentifier></calledDevice><lastRedirectionDevice><notRequired
/></lastRedirectionDevice><localConnectionInfo>alerting</localConnectionInfo><cause>normal</cause></DeliveredEvent>
Figure 5.15
3. MCM Looks up the calling DN number.
Here, as depicted in figure 5.15, the MCM tries to perform the lookup for the calling
number. The Signalling Server already has the E.164 telephone format for the Called
Number. It is part of the EPID generated during the telephone acquisition process by the
10. OCS client. There is no impact on the message routing even if there is no match for the
Calling Number. The purpose is only to replace the calling number with the E.164 format
number. (A)
Instance-Id: 000001C2
Direction: outgoing
Peer: 47.164.113.137:5060 OCS FE IP address
Message-Type: request
Start-Line: INFO sip:usera@demolab.com;opaque=user:epid:kI5Vslm4gl2F2-3nWKKnhQAA;gruu
SIP/2.0
From: <sip:4115@demolab.com>;tag=19042ee0-6a71a42f-13c4-40030-1f8cde-706ef19f-1f8cde
To: “User A"<sip:usera@demolab.com>;tag=b0434a26b8;epid=f60275d516
CSeq: 29 INFO
Call-ID: b094883980e94e0490ed9174a99f481d
Via: SIP/2.0/TCP 47.164.113.136:2013;branch=z9hG4bKE43CDFCD.E8DF3A61;branched=TRUE
Figure 5.16
4. The OCS Application Proxy Server forwards the INFO message to the OCS Front
End Server
Instance-Id: 00000B46 OC client PC IP address
Direction: outgoing
Peer: 47.164.154.177:1219
Message-Type: request
Start-Line: INFO sip:47.164.154.177:1219;transport=tcp;ms-opaque=eb89915611;ms-received-cid=4C00;grid SIP/2.0
From: <sip:4115@demolab.com>;tag=19042ee0-6a71a42f-13c4-40030-1f8cde-706ef19f-1f8cde
To: “User A"<usera@demolab.com>;tag=b0434a26b8;epid=f60275d516
CSeq: 29 INFO
Call-ID: b094883980e94e0490ed9174a99f481d
Via: SIP/2.0/TCP 47.164.113.137;branch=z9hG4bKD325EF8D.891B80DD;branched=FALSE;ms-
internalinfo="abZRshPHdmZ78n8AVeVQlVIBFAXimJG4DdfUZOigAA"
Authentication-Info: Kerberos
rspauth="602306092A864886F71201020201011100 FFFFFFFF894FCA2C25064B8E6455B9E01309F83A", srand="41489DF8",
snum="77", opaque="76A291EA", qop="auth", targetname="sip/CtfOcsFeSrv1.ctflab.net", realm="SIP Communications
Service"Max-Forwards: 68
Via: SIP/2.0/TCP 47.164.113.136:2013;branch=z9hG4bKE43CDFCD.E8DF3A61;branched=TRUE;ms-received-
port=2013;msreceived-cid=5000 Content-Length: 688
Via: SIP/2.0/TCP 47.164.113.106:5060;branch=z9hG4bK-1f93be-7b591ea8-55d0a836;ms-received-port=3597;ms-receivedcid
Figure 5.17
5. The OCS Front End Server sends the INFO to the Office Communicator user Client
PC
This per-hop basis call walkthrough shows how the various messaging is interpreted amongst the
servers and signaling systems that make up this solution, to enable a converged office call to be
made successfully in fully converged environment.
11. As we also demonstrated in the live demo, three basic call features - Call Transfer, Call on Hold
and Call Forward were also tested on the client. This was to determine if the expected behavior
associated the telephony system would also be mirrored on the OCS client.
There a number of configuration parameters required for this integrated environment to work in
an intuitive fashion – and therefore be user friendly. PBX Integration has to be marked in Active
Directory, in addition to the Server URI line being filled in with the appropriate user details e.g
(userA@testdemo.com) as well as the user having activated Phone Integration on their Office
Communicator client. Also, the Address Book Service in the Windows environment has to be set
up correctly for each Office Communicator client.
This results in the user receiving one, incoming merged pop-up toast on the PC, with the correct
CLID (Calling Line Identification Display) details presented of the incoming call.
The omission or incorrect input of any of these values will result in either a dual pop-up being
presented (one for each device – with no indication of which pertains to what device) as well as
the incorrect, or no CLID at all, being shown
In this scenario, there will be two possibilities for an incoming call depending on the Preferred
Calling Device value chosen by the user:
If the user has selected, and then also set their Preferred Calling Device to be Computer,
one incoming call notification pop-up is received, with an option to Redirect to phone as
the answering device.
If the user has selected, and then also set their Preferred Calling Device to be Phone,
again, one incoming call notification pop-up is received, with an option to Redirect to
communicator as the answering device
Figure 5.4
12. As illustrated by the screenshots in Figure 5.4, both types of user defined settings, whether set to
accept calls on the phone or the OCS client, both only receive one incoming call pop-up
notification with the correct CLID indicated, as well as being offered the option of redirecting
the call to the other device (or alternative devices – such as the mobile phone) if required.
This is true of all the call scenarios attempted, as described earlier. The call features of Call
Transfer, Call on Hold and Call Forward previously mentioned also worked as expected, even
though there was no provision for these features on the PBX, they were purely set from the PC
based, and user controlled client.
A call initiated from any device, whether an internal user on a deskphone or an OCS client, or an
externally generated call from a mobile phone or landline, all had the same result in that the user
of the Nortel Converged Office Solution received uniform notifications according to what they
had individually configured as their preferred communications device.