A trunk is a communications line or link designed to carry multiple signals simultaneously to provide network access
between two points. Trunks typically connect switching centers in a communications system. The signals can convey any type
of communications data.
As for the difference between Trunks and Access ports, a trunk does add dot1q or ISL tags directly to frames and can exist
on all or multiple vlans. While an access port only passes traffic from a set vlan but does not modify the frame with a
vlan tag.
uccm unified communications call manager
unity voicemail
VoIP
call center
routine security and performance audits on networks and recommends improvements
Creates and maintains network and project documentation
network design review and changes to meet strategic business needs
BGP,
EIGRP
Enhanced Interior Gateway Routing Protocol (EIGRP) is an advanced distance-vector routing protocol that is used on a
computer network for automating routing decisions and configuration. The protocol was designed by Cisco Systems as a
proprietary protocol, available only on Cisco routers. Partial functionality of EIGRP was converted to an open standard in
2013[1] and was published with informational status as RFC 7868 in 2016.
EIGRP is used on a router to share routes with other routers within the same autonomous system. Unlike other well known
routing protocols, such as RIP, EIGRP only sends incremental updates, reducing the workload on the router and the amount of
data that needs to be transmitted.
EIGRP replaced the Interior Gateway Routing Protocol (IGRP) in 1993. One of the major reasons for this was the change to
classless IPv4 addresses in the Internet Protocol, which IGRP could not support.
HSRP Hot Standby Router Protocol , STP, SIP,
MGCP
The Media Gateway Control Protocol (MGCP) is a signaling and call control communications protocol used in voice over IP
(VoIP) telecommunication systems. It implements the media gateway control protocol architecture for controlling media
gateways on Internet Protocol (IP) networks connected to the public switched telephone network (PSTN).[1] The protocol is a
successor to the Simple Gateway Control Protocol (SGCP), which was developed by Bellcore and Cisco, and the Internet
Protocol Device Control (IPDC).[2]
The methodology of MGCP reflects the structure of the PSTN with the power of the network residing in a call control center
softswitch which is analogous to the central office in the telephone network. The endpoints are low-intelligence devices,
mostly executing control commands from a call agent or media gateway controller in the softswitch and providing result
indications in response. The protocol represents a decomposition of other VoIP models, such as H.323 and the Session
Initiation Protocol (SIP), in which the endpoint devices of a call have higher levels of signaling intelligence.
MGCP is a text-based protocol consisting of commands and responses. It uses the Session Description Protocol (SDP) for
specifying and negotiating the media streams to be transmitted in a call session and the Real-time Transport Protocol (RTP)
for framing the media streams.
MPLS
Multiprotocol Label Switching (MPLS) is a routing technique in telecommunications networks that directs data from one node
to the next based on short path labels rather than long network addresses, thus avoiding complex lookups in a routing table
and speeding traffic flows.[1] The labels identify virtual links (paths) between distant nodes rather than endpoints. MPLS
can encapsulate packets of various network protocols, hence the "multiprotocol" reference on its name. MPLS supports a
range of access technologies, including T1/E1, ATM, Frame Relay, and DSL.
, DMVPN dynamic multipoint
VPNs traditionally connect each remote site to the headquarters; the DMVPN essentially creates a mesh VPN topology. This
means that each site (spoke) can connect directly with all other sites, no matter where they are located.
A DMVPN service runs on VPN routers and firewall concentrators. Each remote site has a router configured to connect to the
company’s headquarters VPN device (hub), providing access to the resources available. When two spokes are required to
exchange data between each other -- for a VoIP telephone call, for example -- the spoke will contact the hub, obtain the
necessary information about the other end, and create a dynamic IPsec VPN tunnel directly between them.
Example network diagram of a dynamic multipoint VPN
DMVPN diagram
Direct spoke-to-spoke deployments provide a number of advantages when compared to traditional VPN deployments:
Traffic between remote sites does not need to traverse the hub (headquarter VPN router).
A DMVPN deployment eliminates additional bandwidth requirements at the hub.
DMVPNs eliminate additional network delays.
DMVPNs conserve WAN bandwidth.
They lower costs for VPN circuits.
They increase resiliency and redundancy.
DMVPN deployments include mechanisms such as GRE tunneling and IPsec encryption with Next Hop Resolution Protocol (NHRP)
routing that are designed to reduce administrative burden and provide reliable dynamic connectivity between sites. It is in
every company’s advantage to make use of DMVPN where possible, to help reduce WAN costs and increase bandwidth and
reliability.
, QoS In the field of computer networking and other packet-switched telecommunication networks, quality of service refers
to traffic prioritization and resource reservation control mechanisms rather than the achieved service quality. Quality of
service is the ability to provide different priority to different applications, users, or data flows, or to guarantee a
certain level of performance to a data flow.
network security, analyzing security risks and developing responses
project management methodologies and ITIL Service Operation concepts
cisco finesse
emergency responder
expressway
Cisco Expressway offers users outside your firewall simple, highly secure access to all collaboration workloads, including
video, voice, content, IM, and presence. Collaborate with people who are on third-party systems and endpoints or in other
companies. Help teleworkers and Cisco Jabber mobile users work more effectively on their device of choice.
jabber
solarwinds orion NPM NCM
NTA (network topology mapper)
VNQM (VoIP & Network Quality Manager)
UDT (User Device Tracker)
SIP trunks
Session Initiation Protocol (SIP) is a communications protocol that is widely used for managing multimedia communication
sessions such as voice and video calls. SIP, therefore is one of the specific protocols that enable VoIP. It defines the
messages that are sent between endpoints and it governs establishment, termination and other essential elements of a call.
SIP can be used to transmit information between just two endpoints or many. In addition to voice, SIP can be used for
video conferencing, instant messaging, media distribution and other applications. SIP has been developed and standardized
under the auspices of the Internet Engineering Task Force (IETF).
In short, if you want to an all-inclusive solution to your business communication needs, SIP trunking it is your best bet.
Employing further tools, such as Asterisk, will make your SIP communications platform even better because you can customize
it to your needs.
SIP trunking delivers telephone services and unified communications to customers with SIP-enabled PBX and unified
communications solutions. In this case, call management, voicemail, auto attendants and other services are provided by the
PBX. The SIP trunks provide the connection between the PBX and the public telephone network, replacing the need for legacy
telephone lines or PRIs (Primary Rate Interface). This gives businesses the ability to select the IP-PBX hardware and
software that works best for them, while freeing them from the expense and inflexibility of traditional phone lines and
carrier relationships.
SIP trunking delivers telephone services and unified communications to customers with SIP-enabled PBX and unified
communications solutions. In this case, call management, voicemail, auto attendants and other services are provided by the
PBX. The SIP trunks provide the connection between the PBX and the public telephone network, replacing the need for legacy
telephone lines or PRIs (Primary Rate Interface). This gives businesses the ability to select the IP-PBX hardware and
software that works best for them, while freeing them from the expense and inflexibility of traditional phone lines and
carrier relationships.
The other ways to deploy VoIP are managed and hosted IP PBX. The latter is a hassle-free version where you have a provider
who oversees everything. You don’t have to get the hardware yourself, or set up the SIP trunking, because you’ll be getting
a pre-configured VoIP system. This is ideal for companies that don’t have the capital to put up a fully customized SIP
trunking service. Remember that it involves creating applications and buying hardware, so if you’re not up to doing all of
that, you have the choice of going for a managed IP PBX.
SIP technology, however, is fast becoming the preferred method of deploying VoIP. Among the benefits that indicate how SIP
works better in VoIP are the reduced costs it offers, the augmented efficiency, as well as its scalability compared to
older systems.
So there really is no such thing as SIP vs. VoIP. SIP is an industry standard method of achieving VoIP, but it’s a
preferred deployment method because of scalability. Your company won’t be limited to using voice communication, as you can
expand into video, instant messaging and more. Businesses looking to improve their communications and reduce cost by moving
to VoIP should carefully consider each of the ways it can be deployed, including SIP trunking, and select the one that
provides the greatest benefit for them.
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