Session Initiation Protocol - Essay Example

ESSAY QUESTIONS Essay Questions Sip Trunking Session Initiation Protocol (SIP) may be termed as a signaling communications protocol that may be used for signaling system for disconnection, connecting and monitoring a multimedia, and communications sessions that are voice based, over the internet. SIP is defined by (IETF) Internet Engineering Task Force as “Peer to peer multimedia signaling protocol that combines with other internet services to convey rich communications” (Rosenberg,et. al., 2002). SIP trunking means that streams from an IP media that are broadcasted from within an enterprise remains a stream from an IP Media and passes somewhere else within the enterprise or across the enterprise’s boundary to a different enterprise through Internet protocol. In other words, a Session Initiation Protocol trunk is the digital data transmission linkage with SIP protocol that signals interfaces recognizing SIP media protocols and SIP signals using the RTP (Real Time Protocol) for media (voice transmission) and RTCP (Real Time Control Protocol) and QoS (Quality of Service). This setting decreases the requirement to be dependent on a local telephone system through employing media’s hardware gateways at the enterprise edge and the carrier edge which results in considerable savings for the enterprise. This may often be referred to as the Public Switched Telephone Network (PSTN). Besides this, additional cost savings may be possible through cutting down on the costly telephone desk sets that are normally utilized in most offices across the world through introducing intelligent soft-phones in their place, as well as, eliminating costly SDN PRI (Primary Rate Interfaces)and (BRI)Basic Rate Interfaces. A SIP trunk is a direct connection connecting an enterprise and an ITSP (Internet telephony service provider). It facilitates an enterprise to extend VoIP (voice over IP) telephony beyond an enterprise’s firewall without the necessarily using an IP-PSTN gateway. This configuration is easier and less costly to maintain, design, operate and also upgrade. According to Rez, since ITSPs deliver services (conspicuously long distance) at considerable savings, an enterprise’s investment in SIP trunking may give a substantial and quick ROI (Return On Investment)” (Rez, 2010). SIP trunking facilitates good working environment with long-distance communication; hence, there is less maintenance and fewer management costs. It also facilitates real time services, instant messaging, updates and conferencing. Equipments required for SIP trunking To effectively deploy an SIP trunk the following three equipments may be necessary: The first equipment is a PBX, with an enabled SIP trunk side: the PBX communicates with all end-points over an Internet Telephony and can be both the analog PBX and the modern digital PBX. The only constraint may be that the SIP trunking connectivity interface ought to be available. The second equipment is an Enterprise edge device that can understand SIP: The local area network (LAN) enabled PBX connects to the ITSP through the Enterprise edge equipment. The enterprise edge equipment can be a firewall enabled that has full SIP support or merely an edge device that connects to the firewall and also handles to the SIP traffic’s transversal. The third equipment is an ITSP (Internet Telephony service provider) for SIP trunking: thus connectivity is provided to the PSTN (Public Switched Telephone Network) for landline and cell phones communication. SIP trunking and RFCs RFC’s that are defined for the SIP are its initial RFC-3261, and it has been updated by RFC’s 3853, 3265, 4916, 4320, 5621,5392, 5626, 5922, 5630, 5954, 6141,6026, 6878, 6665 (Rosenberg,et. al., 2002). However, explicitly for SIP trunk, there is no RFC that is clearly mentioned. Nonetheless, the third party element on SIP’s success depends on, are compliant with a number of the above RFCs. Conclusion SIP trunking may be a solution that numerous individuals or enterprises may use depending on its intended use. In some cases, it may be inexpensive to implement it and when it is not, the purchasing party ought to be confident regarding its security. SIP has registration functions that may add locations to proxy-servers and runs on top of a range of transport protocols. RFCs will continue to be made so as to update SIP and keep up with the progression of technology alongside the innovator’s ideas. References Cross, T. (2011). “SIP Trunking in Deploying Lync: Advice.” UC News. Retrieved from, http://www.telecomreseller.com/2011/09/22/sip-trunking-in-deploying-lync-advice/ Rez. (2010). “SIP Trunking: What Is It? Why Do I Need It? How Do I Deploy It?” technet.microsoft.com. Retrieved from http://technet.microsoft.com/en-us/library/ff383371(v=ocs.14).aspx Internet Rosenberg,et. al. (2002). SIP: Session Initiation Protocol. The Internet Society. Retrieved Nov from, http://tools.ietf.org/html/rfc3261 Voice over WLAN (Wireless Local Area Network) Wireless LANs (WLANs) are fast becoming pervasive among large organizations. At present, Wireless LANs are moving into being a critical element in companies’ networks owing to an increase in wireless voice clients. Moreover, enterprises understand the influence of mobile workforce that may maximize productivity. Enterprises may further their business relationship, responsiveness and also save on the overall cost through addition of VOIP in their wireless networks. VOIP requires a faster transmission and quality service when compared to data since there are unique requirements for Voice over WLAN. In the ever changing world, enterprises ought to upgrade for better end-user support so as to remain relevant. Through creating Voice over WLAN, there is an increase in the effective communication to major players in a trade unit. This essay discusses how voice traffic is moved over a typical WLAN (wireless local area network) and also provides information on the benefits to using a wireless network, the drawbacks related with using a wireless network and the best solutions to overcoming these issues. A Voice over WLAN or (Wireless Voice network) is how digital voice traffic may be sent over a wireless broadband network. The kind of wireless set-up may be basically VoIP sent over an Access Point (WiFi). Within the business, this kind of VoIP set-up is refferd to as WiFi VoIP or VoWLAN since it makes use of the transporting data standards for wireless technology, set by the IEEE 802.11. So as to make use of this technology, the use a VoWLAN enabled device may requires; the most familiar device on the main stream market may be the PDA. There may be other devices accessible in the market like the WiFi handset that resembles a cell phone; however, it allows voice data packets to be sent over the network rather than the usual analog voice signal. The major enterprise level devices may not be a device but rather software-based and may be loaded onto a laptop or a workstation. This software is referred to as a soft-phone, and it is a virtual phone which route calls via the WLAN AP to the IP PBX or a VoIP gateway. There are disadvantages and advantages to using this kind of technology. The main advantage is the cost associated since majority of the users ought to have WiFi hotspot available so as to stay connected. “A huge advantage of wireless VoIP may be that IP phones work on Wi-Fi networks may be utilized in place of cell phones in most cases. Hotspots such as public 802.11 are often available or free at a minimal daily cost. When one is connecting to the Wi-Fi network for e-mail or Web access, there is no additional cost to make calls (VoIP calls) except the VoIP service cost that is typically far less than the cell phone service cost and may facilitate free international calling that is unlimited, something that is never got from majority of the cellular plans” (Shinder, 2014). With the technology of WiFi, one can take a device enabled with VoWLAN in a place where there is a service; thus saving on costs. However, there are also some disadvantages. The Voice over WLAN technology may only be on the ground floor. This technology does not have the capability to have a regular connection to the network devoid of a WiFi connection like a normal cellular device. Another major disadvantage may be network security. Security of whichever network is at all times of great importance, the Voice over WLAN technology may still be vulnerable to intruders or outside attacks. According to computerweekly.com, security may be a vital concern in any deployment that is wireless; however, VoWLAN creates new-fangled vulnerabilities. Numerous handsets that are currently available in the market support only the WEP security standard that has proven to provide insufficient protection against outside attacks and intruders. A Wired Equivalent Privacy (WEP) password may be cracked by intruders simply to make free phone calls at the enterprise’s expense; however, the aim may also be to access and sabotage corporate data in the enterprise” (Understanding VoWLAN, 2014). A possible way to overcome these problems may be to implement the 802.11i standard that provides a yet-to-be hacked encryption and authentication solution. In parallel, the Wi-Fi Protected Access (WPA) may be implemented to ensure that all devices that use the VoWLAN will support the strongest probable security standard (Kodavarti, 2003). Conclusion The important information regarding the Voice over WLAN technology may be that it is a new but innovative technology within the industry. It may or may never work for every organization; however, it is already doing well in the medical field. Most hospitals have already deployed VoWLAN in majority of the major cities. This facilitates quick communications by doctors and nurses which is a key requirement in this kind of field. Voice over WLAN is a technology that is here to stay, whether it remains in its current form or is changed. With time just like with Wi-Fi itself it may only be enhanced. References Kodavarti, R. (2003). “Overcoming QoS, Security Issues in VoWLAN Designs.” Texas Instruments. Retrieved from http://www.eetimes.com/document.asp?doc_id=1271889 Shinder, D. (2014). “10 things you should know about VoIP over wireless.” www.techrepublic.com. Retrieved from http://www.techrepublic.com/blog/10-things/10-things-you-should-know-about-voip-over-wireless/ “Understanding VoWLAN.” (2014). www.computerweekly.com. Retrieved from http://www.computerweekly.com/feature/Understanding-VoWLAN Real-time Transport Protocol (RTP) RTP is an acronym for Real-time Transport Protocol that defines a standard packet format for delivering video and audio over the internet (What is RTP (Real-time Transport Protocol)?, 2014). Its definition lies in the RFC 1889 and was developed by the AVTW (Audio Video Transport Working) group and was first published in the year 1996. Real-time Transport may be used extensively in entertainment and communication systems that engage in streaming media; for instance, web-based “push-to-talk” features telephony, television services and video teleconference applications. RTP is an end-to-end real-time transmission of data, meaning that data may be sent to the client(s) from the server and may be in actual time. Picture something like the internet radio that is audio only, but there ought to be few lapses in the transmission because the data is typically buffered. When there is storage of data in memory and this data is played from the memory, this is referred to as buffering. The buffer allows for a few seconds of stored information prior to playing. Several applications may permit setting of the buffer amount. Real-time Transport Protocol is used in combination with the RTCP (RTP Control Protocol). Whereas the Real-time Transport Protocol carries the media streams; for instance audio and video media streams, the RTP Control Protocol is used to monitor quality of service (QoS) and transmission statistics and aids in synchronization of several streams. Real-time Transport Protocol is originated and received on port numbers that are even, and the associated RTP Control Protocol communication uses the subsequently higher odd port number. Real-time Transport Protocol may be termed as one of the foundations of VoIP, and it may be used in combinations with SIP that assists in setting-up the connections in the network. Initially specified in IETF (Internet Engineering Task Force), the RTP is an Internet protocol standard that specifies a method for programs to direct the real-time multimedia data transmission over either multicast or unicast network services. Real-time Transport Protocol was designed by the Audio-Video Transport Working Group in IETF to support video conferences with numerous, geographically separated participants (Rouse, n.d). Real-time Transport Protocol is normally used in applications that involve Internet telephony. Nonetheless, RTP never in itself, assure real-time multimedia data delivery given that, it is dependent on the characteristics of a network; however, it does provide the means to manage the data when it arrives, to satisfactory effect. Components for RTP The first RTP component is a sequence number that may be used for lost packets detection. The second RTP component is the payload identification that describes the explicit media encoding so that it may be changed stipulated that it has to adapt to a bandwidth variation. The second RTP component is frame indication that marks the each frame’s beginning and end. The second RTP component is source identification that identifies the frame originator. Finally, the fifth RTP component is intramedia synchronization that applies timestamps to sense dissimilar delay jitter within a single stream and recompense for it (Rouse, n.d). On the other hand, RTPC components include: first, QoS (quality of service) feedback that includes the round-trip time, jitter and the numbers of lost packets so that the sources can accordingly regulate their data rates. The second RTPC component is the session control that makes use of the RTCP BYE packet to permit participants to point out that they may be leaving a session. The second RTPC component is identification that includes a participants telephone number; name and e-mail address so as inform other participants. The last RTPC component is inter-media synchronization that enables the synchronization of video and audio streams transmitted separately (Rouse, n.d). RTP and RFCs Specified in the RFC-2509, the CRTP (Compressed RTP) was developed to reduce the size of the RTP, IP and UDP headers. Nevertheless, it was designed to work with fast point-to-point and reliable links. Under less optimal circumstances, where there may be out-of-sequence packets, long delays and packet loss, the CRTP never functions well for VoIP (Voice over IP) applications. An additional adaptation may be the ECRPT Enhanced (CRPT) that was defined in a consequent Internet Draft document to conquer that problem. Conclusion RTP provides identification of the source, timing of information and the payload type for applications. In addition, the associated control protocol, RTCP, provides information regarding the perceived QoS (Quality of Service). As a result, the RTP protocol appears to suite the real-time traffic delivery pretty well. References Rouse, M. (n.d). “Real-Time Transport Protocol (RTP).” searchnetworking.techtarget.com. Retrieved from, http://searchnetworking.techtarget.com/definition/Real-Time-Transport-Protocol “What is RTP (Real-time Transport Protocol)?” (2014). www.3cx.com. Retrieved from, http://www.3cx.com/pbx/rtp/ VoIP over Internet Nowadays, there may be a few different techniques that one can use to carry one’s VoIP services. These include using MPLS, private lines, Frame-relay based WANs, or basically, just the internet. This topic discusses the problems that be encountered when using the internet to carry one’s VoIP service, and also outlines the methods one can apply to overcome these problems. Quite a number of years, we have observed the advancements in technology to offer packet switched communications using IP networks and the Internet. Voice over Internet Protocol also referred to as IP Telephony may be defined as the broadcasting of voice signals over private data network s or the public Internet. Voice over Internet Protocol technology connects the data and telephony worlds; thus permitting voice and data transmissions to be sent across an extensive assortment of networks. This transmission happens when VoIP converts telephone signals that are analog signals into digital signals; hence, transmitting them over the Internet. However, there may be drawbacks that one can encounter when using the internet to carry your VoIP service. The first problem of using the internet to carry your VoIP service may be poor internet connection or bandwidth. This problem may be a key drawback since the service basically rides along on top of TCP/IP that never contains any means of specifying the requirement of a bandwidth; hence, this may be what leads to packet loss. As soon as enough packet loss has happened, it makes it unattainable to hold a conversation for the reason that, the other end may not get the message. Packet loss then leads the receiver or the caller having to wait until the internet traffic decreases so as to re-establish the connection. Waiting for internet traffic to reduce can obviously be very expensive for enterprises that rely on telecommunications to trade. One way to overcome this setback is through allowing an outside person, business, vendor, etc. the right of entry to the enterprises’ extranet. According to Techopedia, “an extranet may be a controlled private network permitting suppliers, customers, vendors, partners and other enterprises to gain information, normally regarding a specific educational institution or company; through doing so without compromising access to the enterprises’ entire network. An extranet may often be a private component of a website which is restricted to pick users through their passwords, user IDs and other mechanisms for authentication on a login page” (Janssen, 2012). In other words, it permits outsiders to have the right of entry to an enterprise’s intranet reliable and securely. This eradicates the problem of packet dropping since one is allowed to have a secure way for VoIP service to transfer and operate VoIP (Voice over Internet Protocol) data packets. A different way to overcome this setback is through ensuring enough download/upload speed from an enterprises’ Internet Service Provider (ISP). By doing so an enterprise can greatly boost call quality due to the extra speed as a result of which voice packets arrive faster at the destination. The second problem that car arise when using VoIP over the internet is jitter. According Bruno Giguere, the term “jitter” refers to “the inconsistency in the packets’ arrival rate from the same flow of data, and abnormal jitter values may unconstructively impact applications that are real-time such as video and VoIP” (Giguere, 2008). One of the best solutions to prevent jitter is to use jitter buffers. These jitter buffers temporarily store arriving packets in order to minimize delay variations, and if the packets end up arriving past a certain amount of time, then they are discarded. The way to prevent them from arriving too late is to implement a low-latency queue for the VoIP and video traffic. You implement this on various network interfaces which provide you with control over these types of configurations. Conclusion There are two major problems you can encounter when using VoIP over the internet. These problems are bandwidth, and jitter. Insufficient bandwidth can cause poor call quality because the packets aren’t arriving at their destination fast enough due to the enormous amount of traffic flow going across the internet. Upgrading your internet speed will maximize the quality of your VoIP service over the internet by ensuring the packets arrive there in a timely fashion. The next problem VoIP can come across when using it over the internet is jitter which is the difference in time in which packets arrive at their destination. To overcome this problem you can implement jitter buffers which delay the variations in which the packets arrive. All-in-all, using the internet to carry your VoIP service is perhaps the easiest and cheapest way to go, but it is also the most susceptible to encountering problems. References Giguere, B. (2008). “Testing Jitter and Wander in the Field—Is it Necessary?” www.exfo.com. Retrieved from, http://www.exfo.com/archived/corporate/blog- old/dates/2008/10/testing-jitter-and-wander-in-the-fieldis-it-necessary Janssen, C. (2012). “Extranet.” www.techopedia.com. Retrieved from, http://www.techopedia.com/definition/2401/extranet Cisco website. (2014). “How Voice over IP (VoIP) Works.” www.cisco.com. 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