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¹ Includes local and long distance calls to the US48 states and Canada (excluding NWT) at no additional charge. Excludes any local, state or federal taxes and/or fees.

² Volume discounts available for ports of more than 10 numbers.

Sipp is a performance testing tool for the SIP protocol. Its main features are basic SIPStone scenarios, TCP/UDP transport, customizable (XML-based) sipp - Browse /sipp/3.1 at SourceForge.net. The SIP Forum reached an important milestone in January, 2008 when it formally ratified version 1.0 of the SIPconnect Technical Recommendation with the unanimous approval of the SIP Forum Board of Directors and announced the formation of the SIPconnect v.1.1 Task Group. 1) There are two SIP UA's A and B. 2) A sends INVITE with SDP to B. 3) B responds with 100 and then 180 Ringing(wthout SDP) and adds TO tag to 180. 3) After 180 is received, A sends a RE-INVITE before 200 OK is received (i.e RE-INVITE with all the headers having same value as initial INVITE and higher CSEQ value). Header (P2) increases to 0.14 bar, it will decrease the diferential pressure across all of the adjacent traps to 1.86 bar. Using a sizing program, we can see that a small decrease in diferential pressure of 0.105 bar (1.965 - 1.86), will cause a decrease in low of about 65 Kg/hour. With the same amount of condensate being generated by the. For more information about SIP trunking and doing business with SIP.US check out our FAQs. If you are unsure of how many SIP channels you will need or have any other questions about pricing for SIP.US services, please reach out by contacting us or give us a ring at 800-566-9810.

• All Inbound calls include CNAM service delivery included.
• You are able to set PSTN forward and PSTN backup redirects per DID on the web portal.
• You are able to SET the 15-character CallerID Name (CNAM) value per DID at no charge on the portal.
• Enhanced DIDs include the ability to register a location address for 911 calling (test by dialing 933).

³ 1000 pages send/receive per month included, $0.10 each additional page.

• High reliability – No T.38 or Fax over IP problems!
• Web portal to view all faxes sent and received
• All faxes sent and received over traditional copper lines (PRIs)
• Adapter comes pre-configured and ready for use out of the box

$129.00 for our special AudioCodes HTTPS FAX Analog Telephone Adapter device.

⁴ 500 pages per month send/receive per number included, $0.10 each additional page

• Incoming faxes are sent to your registered email address as a PDF
• Send document attachments from your registered email address as outbound faxes
• View a history of your sent and received faxes on our web portal
• Change your Fax CallerID number directly in our portal
• No Faxing over IP – All Faxes sent and received over traditional TDM circuits (PRI lines)

• 10,000 message plan can be shared across multiple DIDs. Other plans are per-DID only.
• Most, but not all DID telephone numbers on SIP.US can be enabled for SMS-to-Email functionality.
• Billing is calculated on a per-message basis and billed once a month on your rebill day as a separate line item.
• When enabled, incoming SMS messages to the enabled DID will be sent to the email address(s) you specify. You can also 'SEND' SMS messages via email using this service. On the management page, you can enable SMS functionality on your DIDs that support SMS as well as specifying the email address(s) where the messages will be sent. You can also view usage on each SMS-enabled DID.

In order to determine the cost of SIP Trunking for your company, it is important to understand a couple of SIP concepts.

A SIP trunk is simply the connection between your PBX and the Public Switched Telephone Network. Think of it as the container for your SIP channels. While the SIP trunk is necessary, without SIP channels, you cannot make or receive telephone calls on your SIP trunk. A SIP trunk from SIP.US can contain any number of channels, and you can also map any number of DID Telephone numbers to the SIP Trunk. SIP.US does not charge you a monthly fee or a setup fee for your SIP trunk. You can add additional SIP Trunks at any time (common if you have more than one location, a second PBX device, want to segment call detail records, etc.)

You can think of SIP channels as the equivalent to old fashioned telephone lines. You need one SIP channel for every concurrent incoming or outgoing call. (This article will help you determine how many SIP channels your business is likely to need. It is often equivalent to the number of analog lines or T1/PRI channels you are currently using.) Your monthly price for SIP Trunking will depend on the number of channels you require plus the additional services you select, such as telephone numbers (DIDs). Each SIP channel from SIP.US includes unlimited outbound calling to the US48 states and Canada AND unlimited inbound calling to your local numbers. There are no long distance or per-minute fees for inbound local or outbound US48 and Canada calls.

SIP.US does not lock customers into long-term agreements. You can increase or decrease the number of channels or cancel all together at any time using our Control Panel. Cancellation will be effective at the end of your billing cycle. There are no partial-month refunds for unlimited SIP trunks or DID numbers. For more information about SIP trunking and doing business with SIP.US check out our FAQs.

If you are unsure of how many SIP channels you will need or have any other questions about pricing for SIP.US services, please reach out by contacting us or give us a ring at 800-566-9810.

Ready to see it in action?

WELCOME TO THE WEBSITE OF THE

The SIP Forum is an industry association with members from the leading IP communications companies. Its mission: To advance the adoption and interoperability of IP communications products and services based on SIP.

The Forum promotes SIP as the technology of choice for the control of real-time multimedia communication sessions throughout the Internet, corporate networks, and wireless networks.

The Forum directs technical activities aimed at achieving high levels of product interoperability, provides information on the benefits and capabilities of SIP, and highlights successful applications and deployments.

Each of our Working Groups and Task Groups has their own mailing list, many of which are open to individual, 'Participant' members. Please feel free to join them, or to join the general 'discussion' mailing list.

The SIP Forum is currently engaged in the following technical activities – each of which has its own Task Group that includes a unique mailing list, chairperson(s) and group of contributors:

The Forum is open to individual 'Participant' members, Academic/Institutional Members, and to corporate 'Full Members'. Individual 'Participant' and Academic/Institutional membership is free.

• View our current Full (Corporate) Member Listing.
• View our Academic Member Listing.
• Find out about Membership.
• View the latest SIP Forum News.
• Need more information or logos? Visit our Press Room.

On-Demand Viewing of SIPNOC 2020 Webinars Now Available!

Presented by the SIP Forum, SIPNOC 2020 was the 10th Annual SIP Network Operators Conference – an event that has earned high praise from attendees for its educational, non-commercial and technical content that focuses on the real-world challenges service providers face when deploying SIP-based services such as STIR/SHAKEN.

Like the SIP Forum's June 2019 STIR/SHAKEN VIRTUAL SUMMIT – SIPNOC 2020 was a virtual event consisting of a special Free-to-Attend series of webinars presented by the experts leading the development of the STIR/SHAKEN Call Authentication Framework, as well as leading companies providing STIR/SHAKEN solutions and deployment expertise.

To register for and view the SIPNOC 2020 Webinar recordings, please visit the official SIPNOC 2020 Agenda and Schedule webpage.

For more information, please visit the SIPNOC 2020 event overview webpage.

On-Demand Viewing of the Special Webinar Series Focused on STIR/SHAKEN Deployment and Enforcement Now Available!

This mid-year virtual event, which ran the week of June 22, 2020, consisted of a special series of free-to-attend webinars presented by the experts leading the development of the STIR/SHAKEN Call Authentication Framework, as well as leading companies providing solutions and deployment expertise.

Visit the STIR/SHAKEN VIRTUAL SUMMIT event webpage for more information.

TRACED Act Signed into Law; FCC Mandates Adoption of STIR/SHAKEN

Developed jointly by the SIP Forum and ATIS (the Alliance for Telecommunications Industry Solutions) to efficiently implement the Internet Engineering Task Force's (IETF) STIR (for Secure Telephony Identity Revisited) standard, SHAKEN (for Signature-based Handling of Asserted information using toKENs) defines a mechanism to verify the calling number and specifies how it will be transported across communications networks.

Together, STIR/SHAKEN offers a practical mechanism to provide verified information about the calling party as well as the origin of the call — what is known as 'attestation' — for the first time in the network. Giving service providers the tools needed to sign and verify calling numbers makes it possible for businesses and consumers to know, before answering, that the calls they receive are from legitimate parties.

With the recent signing into law of the 'Pallone-Thune Telephone Robocall Abuse Criminal Enforcement and Deterrence Act', or the 'TRACED Act' (S.151), by President Trump on December 30, 2019, there is new urgency surrounding the deployment of STIR/SHAKEN within the telecommunications industry.

In addition, on March 31, 2020, the United States Federal Communications Commission adopted new rules requiring implementation of caller ID authentication using 'STIR/SHAKEN.' These rules will further the FCC's efforts to protect consumers against malicious caller ID 'spoofing,' which is often used during robocall scam campaigns to trick consumers into answering their phones.

The TRACED Act authorizes the Federal Communications Commission (FCC) to issue additional civil penalties on individuals who intentionally violate restrictions on the use of automated telephone equipment (i.e., illegal robocalls and spoofing); and directs the FCC to require voice service providers to offer call authentication technologies (i.e., STIR/SHAKEN) to consumers.

For more information about the NNI Task Force and Charter, and to obtain copies of the completed, ratified STIR/SHAKEN specifications, please visit the NNI Task Force Introduction Page.

'Focus on STIR/SHAKEN'

SIPNOC 2019 Concludes – Was Largest and Most Successful Event to Date!

The ninth annual SIP Network Operators Conference — SIPNOC 2019 — was held in Herndon, VA, December 3-5, 2019.

SIPNOC 2019 was 100% focused on the SHAKEN Call Authentication Framework and the expanding efforts to combat the Robocall epidemic and is being specifically developed for industry stakeholders in the Robocall elimination/mitigation ecosystem, including service providers seeking to deploy new solutions, governmental regulators and agencies, equipment manufacturers, enterprise and government agency contact centers, application providers and data analytics firms.

For detailed, archived event information, please visit the SIPNOC 2019 event webpage.

The SIPNOC conferences attract leading technical and operations personnel from the global carrier community and have earned high praise from attendees for their educational, non-commercial and technical content that focuses on the real-world challenges operators face when deploying SIP services in global IP networks. SIPNOC attendees include telecommunications providers, major backbone operators, interconnect and wholesale solution providers, ISPs, cable operators, wireless network operators as well as large enterprises deploying major SIP initiatives.

The three-day conference has attracted technical leadership from MSOs and carriers from North America, South America and Europe including Accent Communications, ADP, Airespring, Allstream, AT&T, Approved Networks, Armstrong Utilities, babyTel, Baltimore-Washington Telephonone Company, Bandwidth, Bell Canada, Bluetone Communications, Brighthouse Networks, British Telecom, Broadvox, CableOne, Cablevision, Centracom, CDK Global, Centurylink, Cincinnati Bell, Cinchcast, Charter Communications, Cologix, Comcast, Consolidated Telecom, COX Communications, Deutsch Telecom, Dialpad, E Street Communications, Evolve IP, Frontier Communications, Global Crossing, GRUPO GTD, Hargray Communications, iBasis, IDT, IntelliVerse, Lanck Telecom, Level 3, Livevox, LINX America, LogMeIn, Lumos Networks, MarcaTel, Megapath, MTS Allstream, Nextel, Optimum Lightpath, One Access Networks, Orange, Purple Communications, RingCentral, Rogers, SDN Communications, Shentel, Socket Telecom, Soleo Communications, Somos, Sorenson Communications, Sprint, Swisscom, Telefonica, Telmex, TelNet Worldwide, Time Warner Cable, TDS Telecom, Telefonica International Wholesale Services, XConnect Networks, XO Communications, Uni-tel, Verizon, Videotron, Vonage, Voxbone, Vocalocity, WDT, Windstream, and Ziggo.

In addition to carrier participants, SIPNOC has also attracted a myriad of SIP community stakeholders from vendors, governments and research organizations such as Alorica, Amazon, Apple, ATIS, AudioCodes, Avaya, BlueAlly, Broadsoft, CableLabs, California State University, Canadian Radio-Television and Telecommunications Commission (CRTC), CaptionCall, Cequint, Cisco Systems, Columbia University, Commetrex, CounterPath, CTIA, Democon, Dialogic Corporation, U.S. Department of Homeland Security, Dialog Technology, DISA (U.S. Defense Information Services Agency), ECG, Edgewater Networks, EGH, Equinox Information Systems, Ericsson, U.S. Federal Communications Commission, First Orion, Frequentis AG, GENBAND, Georgetown University, Hiya, Huawei Technologies, iconectiv, Illinois Institute of Technology, Ingate Systems, ISOC, LogMeIn, LucidTech, Metaswitch, Microsoft, MITRE, Neustar, Netmaker Communications, New York Department of State, NICE Incontact, Noble Systems, Nuance Communications, Numeracle, Penn State University, Polycom, Professional Association for Customer Engagement (PACE), Oracle, Redshift Networks, REVE Systems, Ribbon Communications, Sangoma Technologies, Sansay, SecureLogix, Siemens, snom, Sonus Networks, TelcoBridges, TeleStax, Teraquant, TNSI, Unify, University of New Hampshire Interoperability Laboratory, University of Virginia, U.S. Dept of Transportation, U.S. Government Accountability Office (GAO), USTelecom Association, and Vigilsec.

SIP Forum 2020 Annual General Meeting

The SIP Forum 2020 General Meeting was held on Tuesday, 26 January 2021.

In the meeting, a review of the financial operations was conducted, as well as a review of the operational and technical activities that have occurred over the past 12 months since the last General Meeting. In addition, votes were cast for new board members and for two other voting issues.

View more information about the meeting, including the full text of the 2020 Annual General Meeting Notice.

Two New ATIS/SIP Forum NNI Task Force Resources Available

ATIS and the SIP Forum have released two new technical reports advancing industry efforts to mitigate unwanted robocalling.

The Technical Report on a Framework for Display of Verified Caller ID provides a framework for signaling verified Caller ID information from the network to a User Equipment (UE), and displaying the information on the UE in a uniform manner, independent of technology. The goal is to produce display guidelines that help meet the goals of regulators and consumer protection agencies for empowering consumers with simple and effective information on the displayed Caller ID. This will help consumers know if a call is from who it says it is from so they can make an informed decision on whether to answer. Screenflow 7 7 3 1.

The Technical Report on SHAKEN API for a Centralized Signing and Signature Validation Server provides additional detail on one possible implementation of the SHAKEN specification in service provider networks. It defines a RESTful interface (API) that can be used in the SHAKEN framework to interface with a centralized server to sign and verify telephony identity for multiple nodes within a network. Implementing SHAKEN authentication and verification functions in a centralized server that can be accessed via an API supports cloud-based implementations and is designed to enable a more cost-effective network deployment. Amandine 1 0 78.

All SHAKEN resources are available on the SIP Forum site here.

SIP Forum Ratifies the SIPconnect Technical Recommendation Version 2.0!

The SIP Forum reached an important milestone by formally ratifying Version 2.0 of the SIPconnect Technical Recommendation on November 28, 2016, with the unanimous approval of the SIP Forum Board of Directors.

The formal adoption by the SIP Forum Board of the SIPconnect 2.0 Technical Recommendation is based on recognition that the recommendation has been through credible peer review, including broad membership and significant community review, that it is stable and is well-understood, and that it is believed to have resolved known design choices.

The SIPconnect 2.0 Technical Recommendation is a profile of the Session Initiation Protocol (SIP) and related media aspects that enables direct connectivity between a SIP-enabled Service Provider Network and a SIP-enabled Enterprise Network. It specifies the minimal set of IETF and ITU-T standards that must be supported, provides precise guidance in the areas where the standards leave multiple implementation options, and specifies a minimal set of capabilities that should be supported by the Service Provider and Enterprise Networks.

SIPconnect 2.0 effectively extends SIPconnect 1.1. Where SIPconnect 1.0, and 1.1 focused primarily on basic network registration, identity/privacy management, call originations, call terminations, and advanced services, the 2.0 version adds additional guidance on Security, Emergency Calling, and IPv6.

Where appropriate, recommendations from SIPconnect 1.1 have been left unchanged, although some modifications to prior recommendations have been made based on experience and feedback gathered through adoption of SIPconnect 1.1 in the industry.

View the full text of the SIP connect 2.0 announcement.

View or download the ratified SIPconnect Technical Recommendation Version 2.0.

View more information about the SIPconnect 2.0 Task Group charter.

SIP Forum Launches SIPconnect Certification Testing Program

We are proud to announce the launch of the SIPconnect Certification Testing Program, a unique certification testing program that includes a new certification test suite and test platform, as well as an associated 'SIPconnect Certified' logo program that will serve as the official 'seal of certification' for companies products and services that have successfully passed the certification test and officially achieved conformance with the SIPconnect specification.

The result of three years of planning and development, the new SIPconnect Certification Testing Program will be hosted by the UNH-IOL, a well-known independent, third-party laboratory dedicated to broad-based testing and standards conformance services for networking industries. The program will be available to IP communications equipment vendors, including IP-PBX and SBC vendors, and service providers who are offering SIP trunking services.

For more information, please read SIPconnect Certification Program Overview webpage.

SIP Forum's SIP over IPv6 Task Group Achieves Milestone with the Publication of 'Interoperability Impacts of IPv6 Interworking with Existing IPv4 SIP Implementations'

The continued proliferation of IPv6 infrastructure deployments has resulted in more IPv6 Session Initiation Protocol (SIP) User Agents (UAs) being turned up on networks around the world. Considering the large installed base of IPv4 SIP UAs deployed prior to the deployment of IPv6, it is a well-known fact that not all IPv4 SIP UAs have taken into account all possible IPv4 SIP-to-IPv6 SIP interoperability considerations at the time of their development.

The SIP Forum has announced that the SIP over IPv6 Task Group has achieved a significant milestone in its mission to improve the interoperability of IPv6 Interworking with IPv4 implementations with the publication of a new document entitled 'Interoperability Impacts of IPv6 Interworking with Existing IPv4 SIP Implementations'.

View the full text of the announcement.

The SIP Forum's SIP over IPv6 Task Group (IPv6) was formed to address key deployment and interoperability issues in the telecommunications industry's migration to SIP over IPv6. The task group, which includes key stakeholders from the service provider, application developer and equipment communities, has developed and ratified a charter with the mandate to identify issues with SIP over IPv6 and assess the impact of transition technologies and dual stack devices on existing SIP networks.

For more information, please visit the SIP over IPv6 Task Group Charter webpage.

SIPit 32 Post-Event Info

SIPit 32 was held at the University of New Hampshire Interoperability Laboratory September 12-16, 2016. A Summary of test results from the event will be available soon.

In addition to the usual thorough testing, SIPit 32 focused on new evolutions in SIP Security such as the mechanisms defined in the STIR working group. The event also conducted tests designed to inform the new SIPBRANDY effort in the IETF.

Read the full SIPit 32 announcement.

View the results of past SIPit events in the SIPit section of the SIP Forum website.

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SIP Forum and BITKOM Partner in the Ongoing Development of SIPconnect

The SIP Forum announced an ongoing alliance with BITKOM, the Federal Association for Information Technology, Telecommunications and New Media in Germany, in the development of updates to the SIP Forum's SIPconnect Technical Specification (currently SIPconnect 2.0).

The formal endorsement of SIPconnect by BITKOM provides valuable technical input and peer review by one of the European Union's most prominent technical associations, and will help ensure that the next versions of SIPconnect meet the requirements of BITKOM members. As part of the agreement, the two organizations will pool requirements as they work towards an update of their respective SIP trunking specifications.

Read the full text of the SIP Forum and BITKOM Announcement.

SIP Forum's Fax over IP Task Group Achieves Milestone with the Publication of IETF RFC 6913

The SIP Forum announced its Fax over IP (FoIP) Task Group has achieved a significant milestone in its mission to improve international IP fax transport services with the publication of RFC 6913 – a new Internet Engineering Task Force (IETF) RFC that introduces a new 'sip.fax' media feature tag that aims to enable the intelligent routing of International faxes and greatly improve the reliability of International faxing services.

RFC 6913, co-authored by David Hanes, Kevin Fleming and Gonzalo Salgueiro, defines and registers with IANA a new 'fax' media feature tag for use with the Session Initiation Protocol (SIP). Currently, fax calls are indistinguishable from voice calls at call initiation. Consequently, fax calls can be routed to SIP user agents that are not fax capable. A 'fax' media feature tag implemented in conjunction with caller preferences allows for early advertisement of fax capabilities and consequently, more intelligent fax call routing.

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See the full text of the RFC 6913 announcement.

As part of the work leading up to this new RFC, the Fax-over-IP Interoperability Task Group published a T.38 SIP-SDP subgroup Problem Statement that details a number of SDP offer/answer interoperability issues found while implementing and connecting T.38 compliant endpoints together, primarily over the SIP signaling mechanism. View or download this document.

In addition, the Forum previously published an official task group problem statement, a document that details the various interoperability issues that currently plague FoIP services. This document is available for download.

The FoIP Task Group has developed a task group charter for viewing.

To view task group documents and other related materials, please visit the FoIP Task Group document repository.

SIP Forum UA Configuration Recommendation Ratified and Published as RFC 6011 by the IETF

The SIP Forum's User Agent Configuration Recommendation for the locating, retrieving and maintaining of SIP User Agents has been ratified and published as RFC 6011 by the Internet Engineering Task Force (IETF).

The publishing of the recommendation as RFC 6011 marked a significant milestone for the SIP Forum and its UA Configuration Task Group, at the time led by Chairman John Elwell, former Head of Standardization Strategy at Siemens Enterprise Communications GmbH. The UA Configuration Task Group had been working on the procedure since 2009, and designed the standard in a manner that addressed the needs of end users and services providers, as well as both small businesses and large enterprises deploying SIP-enabled endpoints.

Now published as RFC 6011 by the IETF's Internet Engineering Steering Group (IESG) after a public review process by the IETF community, the User Agent Configuration Recommendation (UA Config) sets a standard procedure for how a SIP User Agent locates, retrieves and maintains current configuration information for a given SIP Service Provider. It requires that each User Agent, the configuration agent at the service provider and network infrastructure meet such requirements to ensure communication.

View the full text of the announcement.

To view the published text of RFC 6011, please visit http://tools.ietf.org/search/rfc6011.

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