1. Michael Ferens for TCOM500 VoIP Voice over Internet Protocol Definition Definition Emergency Call Services Emergency Call Services

2. Michael Ferens for TCOM500 VoIP Digitalizes and transfers voice through IP packets via the Internet Digitalizes and transfers voice through IP packets via the Internet Uses TCP/IP to travel from point A to Z over the network Uses TCP/IP to travel from point A to Z over the network Uses a gateway for receiving voice packet transmissions from users and then routes them to others, using a carrier system interface sending them over the PSTN Uses a gateway for receiving voice packet transmissions from users and then routes them to others, using a carrier system interface sending them over the PSTN ITU-T H.323 – most common standard being used for purposes of vendor interoperability ITU-T H.323 – most common standard being used for purposes of vendor interoperability Will be available via alternative wireless networks such as WiMAX and Wi-Fi Will be available via alternative wireless networks such as WiMAX and Wi-Fi

3. Michael Ferens for TCOM500 VoIP  How it works: When caller picks up the handset, the off-hook condition signal is sent to the VoIP signaling application using the Cisco Catalyst 4224 Access Gateway Switch When caller picks up the handset, the off-hook condition signal is sent to the VoIP signaling application using the Cisco Catalyst 4224 Access Gateway Switch The session application issues a dial tone and waits for the caller to dial a telephone number The session application issues a dial tone and waits for the caller to dial a telephone number The number is dialed and the session application stores the dialed digits The number is dialed and the session application stores the dialed digits The telephone number is mapped to an IP host via the dial plan mapper. The IP host has a direct connection to either the destination telephone number or a private branch exchange (PBX) The telephone number is mapped to an IP host via the dial plan mapper. The IP host has a direct connection to either the destination telephone number or a private branch exchange (PBX) H.323 session protocol establishes transmission and reception channels for each direction over the IP network. If the call is being handled by a PBX, the PBX forwards the call to the destination telephone. If Resource Reservation Protocol (RSVP) has been configured, RSVP reservations are put into effect to achieve the desired quality of service (QoS) over the IP network H.323 session protocol establishes transmission and reception channels for each direction over the IP network. If the call is being handled by a PBX, the PBX forwards the call to the destination telephone. If Resource Reservation Protocol (RSVP) has been configured, RSVP reservations are put into effect to achieve the desired quality of service (QoS) over the IP network

4. Michael Ferens for TCOM500 VoIP The coder-decoder compression schemes are enabled for both ends of the connection using Real-Time Transport Protocol/User Datagram Protocol/Internet Protocol (RTP/UDP/IP) The coder-decoder compression schemes are enabled for both ends of the connection using Real-Time Transport Protocol/User Datagram Protocol/Internet Protocol (RTP/UDP/IP) Any call-progress indications (or other signals that can be carried inband) are cut through the voice path as soon as an end-to-end audio channel is established. Signaling that can be detected by the voice ports is also trapped by the session application at each end of the connection. Signaling carried over the IP network is encapsulated in Real-Time Transport Control Protocol (RTCP) using the RTCP application-defined (APP) extension mechanism. Any call-progress indications (or other signals that can be carried inband) are cut through the voice path as soon as an end-to-end audio channel is established. Signaling that can be detected by the voice ports is also trapped by the session application at each end of the connection. Signaling carried over the IP network is encapsulated in Real-Time Transport Control Protocol (RTCP) using the RTCP application-defined (APP) extension mechanism. When either person hangs up the phone, RSVP reservations are torn down (if RSVP is used) and the session ends. Each end becomes idle, waiting for the next off-hook condition to trigger another call setup. When either person hangs up the phone, RSVP reservations are torn down (if RSVP is used) and the session ends. Each end becomes idle, waiting for the next off-hook condition to trigger another call setup.

5. Michael Ferens for TCOM500 VoIP

6. VoIP Methods: ATA ATA  Analog Telephone Adaptor is an analog to digital converter and allows use of a standard phone IP Phones IP Phones  IP Phones provided by VoIP that appear as a normal phone but use an Ethernet connector instead of the standard RJ-11 connector Computer to Computer Computer to Computer  Simplest and cheapest method that requires a standard software (microphone, speakers, and sound card)

7. Michael Ferens for TCOM500 VoIP Features (extra charges) Caller ID Call waiting Call transfer Repeat dial Return call Three-way calling

8. Michael Ferens for TCOM500 VoIP US Subscribers: 2004985,000 2005 2.26 to 2.5 million 2006 6.3 million 2009 27 million

9. Michael Ferens for TCOM500 VoIP Advantages: Brings reliable dial tone and all the features a telephone user expects including direct national and international dialing, operator and directory assistance, emergency calling, conference calls, and voicemail Brings reliable dial tone and all the features a telephone user expects including direct national and international dialing, operator and directory assistance, emergency calling, conference calls, and voicemail A VoIP customer can use their ATA anywhere in the world where an Internet connection is available, and the customer’s number will follow A VoIP customer can use their ATA anywhere in the world where an Internet connection is available, and the customer’s number will follow Carrier-grade sound quality Carrier-grade sound quality No FCC tax regulation (yet) No FCC tax regulation (yet) Avoids the tolls charged by ordinary telephone service Avoids the tolls charged by ordinary telephone service Less expensive Less expensive

10. Michael Ferens for TCOM500 VoIP Disadvantages: During times of congestion (data or voice), calls are susceptible to latency, clipping, distortion, echo, delays and other factors that can lower performance and sound quality During times of congestion (data or voice), calls are susceptible to latency, clipping, distortion, echo, delays and other factors that can lower performance and sound quality Inexpensive only when talking to people on the same service. When the calls transmits through other carriers, there can be additional charges Inexpensive only when talking to people on the same service. When the calls transmits through other carriers, there can be additional charges In many locations, LEC’s are “blocking" VoIP from reaching certain SIP addresses for end-user devices In many locations, LEC’s are “blocking" VoIP from reaching certain SIP addresses for end-user devices Must have power on at all times to receive calls Must have power on at all times to receive calls Security – With Internet, packets can be captured meant for someone else Security – With Internet, packets can be captured meant for someone else Emergency 911 calls present a challenge Emergency 911 calls present a challenge

11. Michael Ferens for TCOM500 VoIP The 911 problem: VoIP service simply did not connect to 911 VoIP service simply did not connect to 911 When VoIP service rang to 911 services, it would not include the subscriber’s phone number or location information. When VoIP service rang to 911 services, it would not include the subscriber’s phone number or location information. Some customers claimed that 911 warnings were hidden in pages of Terms and Conditions so they were caught off-guard Some customers claimed that 911 warnings were hidden in pages of Terms and Conditions so they were caught off-guard Some 911 centers do not accept VoIP calls because systems will not indicate caller’s number/address VoIP service did not work during a power outage VoIP service did not work during a power outage VoIP service did not work when the broadband connection (cable modem or DSL) went down or was congested VoIP service did not work when the broadband connection (cable modem or DSL) went down or was congested Federal Regulators ruled that VoIP is not a traditional phone service, therefore, state/local officials can not require VoIP providers to link to 911

12. Michael Ferens for TCOM500 VoIP Solutions: ) In May 19 2005, FCC Chairman Kevin Martin announces that VoIP carriers (only those who access traditional circuit-switched networks) have 120 days to:  Deliver all 911 calls to the local emergency call center  Deliver the customer’s call back number and location information where the emergency call center is capable of receiving it  Inform their customers of the capabilities and limitations of their VoIP 911 service.

13. Michael Ferens for TCOM500 VoIP The Players:  Vonage (32%)  Time Warner Cable – AOL (25%)  Cablevision (18%)  SunRocket  TCS (Telecommunication Systems)  AT&T CallVantage  Broadvox  Lingo  Verizon VoiceWing  Packet8  Skype (Enhanced Service?)  Motorola (Global Venture with Skype)

14. Michael Ferens for TCOM500 VoIP Vonage: Vonage:  Requesting subscribers to complete a form to be eligible for the “E911” service. This information is shared with the local subscriber’s home address so that 911 centers know to possibly follow up or forward your emergency to the local 911 office. TCS TCS  Incorporated Master Street Address Guide (MSAG)-based routing feature in its existing VoIP Positioning Center. This routing feature facilitates the routing of 9-1-1 calls from their VOIP to Public Safety Answering Point (PSAP) emergency call centers. This solution applies to static and nomadic VoIP emergency calls.

15. Michael Ferens for TCOM500 VoIP GETS/WPS: WPS calls can be made to a VoIP subscriber however the priority queuing ends when the transmissions hits the VoIP network Standards work is proceeding to define the Resource Priority Header (RPH) parameter to be used in SIP to indicate a high-priority call