SIP (Session Initiation Protocol) trunking connects a company's PBX to the existing telephone network infrastructure via the internet using VoIP. It was originally designed in 1996 and standardized in 2000. SIP trunking provides benefits like virtual phone numbers, reduced equipment needs, business continuity, and flexible trunk quantities. However, considerations must include system compatibility, additional bandwidth requirements, and challenges like supporting fax/modem traffic and 911 calls. The document discusses ideal environments for SIP trunking like companies with multiple locations, seasonal needs, or those seeking increased functionality. It also reviews cost components and provides an overview of SIP trunking.
A short presentation on SIP Trunking. A background with SIP training and PSTN (T-1, ISDN PRI) and TCP/IP knowledge will be helpful. For more info connect with me at http://TrainingCity.com
The presentation is a compiled assembly from the SIP RFC' s, and original works of Alan Johnston and Henry Sinnreich . It contains Sip Detailed , Call flows , Architecture descriptions , SIP services , sip security , sip programming.
SIP - More than meets the eye
Speakers:
Ofer Cohen - VOIP Group Leader, LivePerson
Yossi Maimon - VOIP Technical Leader, LivePerson
An Introduction to the SIP protocol.
SIP Position in telecommunication networks and the content services.
What is SIP:
The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating sessions consisting of one or several media streams. SIP can be used for two-party (unicast) or multiparty (multicast) sessions. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.
(Source: Wikipedia)
Avaya Aura® Session Border Controller, powered by Acme Packet, secures the IP border for the real time interactive communications that flow outside your internal network. With Avaya Aura® Session Border Controller, your Unified Communications and Contact Center Solutions can securely leverage SIP, while simultaneously extending the power of the Avaya Aura® architecture throughout your enterprise to realize the true benefits of open standards.
A short presentation on SIP Trunking. A background with SIP training and PSTN (T-1, ISDN PRI) and TCP/IP knowledge will be helpful. For more info connect with me at http://TrainingCity.com
The presentation is a compiled assembly from the SIP RFC' s, and original works of Alan Johnston and Henry Sinnreich . It contains Sip Detailed , Call flows , Architecture descriptions , SIP services , sip security , sip programming.
SIP - More than meets the eye
Speakers:
Ofer Cohen - VOIP Group Leader, LivePerson
Yossi Maimon - VOIP Technical Leader, LivePerson
An Introduction to the SIP protocol.
SIP Position in telecommunication networks and the content services.
What is SIP:
The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating sessions consisting of one or several media streams. SIP can be used for two-party (unicast) or multiparty (multicast) sessions. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.
(Source: Wikipedia)
Avaya Aura® Session Border Controller, powered by Acme Packet, secures the IP border for the real time interactive communications that flow outside your internal network. With Avaya Aura® Session Border Controller, your Unified Communications and Contact Center Solutions can securely leverage SIP, while simultaneously extending the power of the Avaya Aura® architecture throughout your enterprise to realize the true benefits of open standards.
If you don't know what SIP is, what it is used for or why you should even care then this is the section for you. This presentation will go over the very basics of SIP and assumes no previous knowledge of SIP or really any other network experience either. Topics it will touch upon is:
- What SIP is all about
- SIP and sessions management
- Basic call flow
- Brief discussion of SIP messages
- SIP and audio
Brief introduction into SIP protocol, how it works, common problems to solve. Tech. details about handshake, SIP Trunks and SIP trunking. Market research.
This presentation and demo show the hardware which consist of 5G UE’s, 5G radios, a fronthaul network and C-RAN with high density switches and servers, a transport network of 3 DWDM switches and a DC network of servers and high density switches. The basic software arrangement will be shown with emphasis on the structure of the orchestration and SDN controllers and the choice of virtualization components and logical networking. An eMBB slice will be brought up which will entail programming of the radios, the fronthaul, backhaul, a node B and the core. Its behavior will be noted through the test UE’s. An URRLC slice will be brought up and its nodeB and core will be demonstrated through its test UE’s showing extremely low latency. An MMTC slice will be brought up and a large number of test IOT devices will be demonstrated via the test UE’s. The eMBB slice will be augmented by programming a slice selection function that will create a ICN slice and an application (TBD) will be shown running over that ICN core (but with the eMBB slice). Spectrum will be reassigned from slice to slice and the changes noted as an optimizer recomputes the proper allocation of resources and executes it. Traffic will be increased and the changes in the backhaul over transport and core function placements will be noted. An additional demonstration will show creation of multiple 4G air interfaces using the same infrastructure network but with 4G radios and 4G UE’s using OAI software and ETTUS SDRs. A Skype session will be created between the two 4G slices. We will also try to show some of the EPC functions being moved while the UE sessions are not impacted.
Author : Peter Ashwood-Smith, Huawei Technologies
Presented at ITU-T Focus Group IMT-2020 Workshop and Demo Day, 7 December 2016.
More details on the event : http://www.itu.int/en/ITU-T/Workshops-and-Seminars/201612/Pages/Programme.aspx
Join us for an introductory webinar on VoIP and learn:
- The fundamental principles of VoIP including RTP and SIP
- What voice metrics to measure and why they matter
- The different methods to monitor and troubleshoot VoIP
If you don't know what SIP is, what it is used for or why you should even care then this is the section for you. This presentation will go over the very basics of SIP and assumes no previous knowledge of SIP or really any other network experience either. Topics it will touch upon is:
- What SIP is all about
- SIP and sessions management
- Basic call flow
- Brief discussion of SIP messages
- SIP and audio
Brief introduction into SIP protocol, how it works, common problems to solve. Tech. details about handshake, SIP Trunks and SIP trunking. Market research.
This presentation and demo show the hardware which consist of 5G UE’s, 5G radios, a fronthaul network and C-RAN with high density switches and servers, a transport network of 3 DWDM switches and a DC network of servers and high density switches. The basic software arrangement will be shown with emphasis on the structure of the orchestration and SDN controllers and the choice of virtualization components and logical networking. An eMBB slice will be brought up which will entail programming of the radios, the fronthaul, backhaul, a node B and the core. Its behavior will be noted through the test UE’s. An URRLC slice will be brought up and its nodeB and core will be demonstrated through its test UE’s showing extremely low latency. An MMTC slice will be brought up and a large number of test IOT devices will be demonstrated via the test UE’s. The eMBB slice will be augmented by programming a slice selection function that will create a ICN slice and an application (TBD) will be shown running over that ICN core (but with the eMBB slice). Spectrum will be reassigned from slice to slice and the changes noted as an optimizer recomputes the proper allocation of resources and executes it. Traffic will be increased and the changes in the backhaul over transport and core function placements will be noted. An additional demonstration will show creation of multiple 4G air interfaces using the same infrastructure network but with 4G radios and 4G UE’s using OAI software and ETTUS SDRs. A Skype session will be created between the two 4G slices. We will also try to show some of the EPC functions being moved while the UE sessions are not impacted.
Author : Peter Ashwood-Smith, Huawei Technologies
Presented at ITU-T Focus Group IMT-2020 Workshop and Demo Day, 7 December 2016.
More details on the event : http://www.itu.int/en/ITU-T/Workshops-and-Seminars/201612/Pages/Programme.aspx
Join us for an introductory webinar on VoIP and learn:
- The fundamental principles of VoIP including RTP and SIP
- What voice metrics to measure and why they matter
- The different methods to monitor and troubleshoot VoIP
Matrix Telecom Solutions: SETU VTEP - Fixed VoIP to T1/E1 PRI GatewayMatrix Comsec
Matrix SETU VTEP is a compact and dedicated gateway for VoIP to T1/E1 PRI network offering high-value communication experience to businesses of all size, Service Providers, Call Centers and simple but cost-effective solution for multi-location branch office communication. This intelligently designed gateway incorporates advanced features with multiple connectivity options to connect with a legacy communication system using T1/E1 or PRI signaling. SETU VTEP offers reliable and cost-effective solutions to the changing requirements of the business communication and offer customer value for money.
Matrix Telecom Solutions: NAVAN CNX200 - Office-in-a-Box Solution for Small B...Matrix Comsec
NAVAN CNX200 is an all-in-one office solution for small businesses and enterprise branch offices with up to 24 users. It combines the functionalities of IP-PBX, Data router, Wi-Fi access point, VoIP-GSM gateway, VPN and Firewall Security in a compact and converged platform. A true office-in-a-box, CNX200 innovates the way small businesses communicate and manage infrastructure, so that they can increase productivity, lower costs and enhance collaboration with customers and suppliers.
SIP servers on embedded systems: Powering SoHo communicationsRADVISION Ltd.
Increasing number of small offices/home offices (SoHo) migrate to VoIP communication based on commodity appliances that provide the connectivity solution.
The appliance is in many cases a combination of:
* Residential gateway (RGW)
* Integrated access device (IAD)
* Key-system/ small PBX
* SIP firewall / GW
In some cases it even can be as part of a set-top box.
All the above is based on SIP server technology that is tailored to embedded devices. This technology involves significant business potential as well as technology challenges that will be elaborated on the webinar.
Webinar de la marca Patton explicando las funcionalidades de los SBC ESBR y escenarios de uso
All IP Networks – EOL for ISDN
Survivability, Service Demarcation, QoS
VoIP Security
VoIP Encryption, SIP TLS SRTP, Network topology hiding
IP Routing Performance
IP Routing Performance for multi service application
Ease of use
Simple configuration, deployment
Pricing
Price-value ratio
4. A SIP (Session Initiation Protocol) connection is a service offered by many ITSP (Internet Telephony Service Providers) that connects a company's PBX to the existing telephone system infrastructure (PSTN) via Internet using the SIP VoIP standard.
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