Sipjs call. Most JS libs focus on SIP over websockets and WebRTC, but in my infrastructure, I do not have WebSockets. If it is an outgoing SIP session Session Initiation Protocol is a critical set of standards to establish real-time digital communication. About. NO_PRACK Server Configuration Guides. Share your screen or desktop. VoIP is an umbrella term for many different forms of voice communication that happen over the internet, and SIP describes exactly how these calls are established, maintained, and disconnected. the Javascript SIP library. session. makeUri() helper to make the URI of the May 24, 2021 · SIP allows people around the world to communicate using their computers and mobile devices over the internet. com Utilize advanced call features such as early media, call hold and resume, and transfers; Send DTMF with SIP INFO; 100% open source, 100% JavaScript; All major browsers supported Create real-time peer-to-peer audio and video sessions via WebRTC. † The transaction number within a single call leg is identified in the CSeq field. Feb 11, 2013 · Easily install & configure Asterisk to work with SIP. remoteStream; // Access local and remote audio tracks. Here's the code of SIP UA registration: uri: '211', Here's the code of call initiation: mediaConstraints: {. Check your PBX to be sure the Twilio IP addresses and ports are allowed. This section of the documentation is intended to help you use SIP. js is a SIP stack for node. js source code to use those. This section details session establishment between two SIP User. It is frequently confused with VoIP calls, but SIP calling uses VoIP to transmit your analog call traffic over the internet. SIP (Session Initiation Protocol) is a signaling protocol used to establish a “session” between 2 or more participants, modify that session, and eventually terminate that session. SIP communication protocol is like a language that allows different devices, such as phones and computers, to Oct 3, 2022 · This process of transmitting voice calls over a SIP channel or trunk is known as SIP calling. 5. In essence, SIP trunking allows for the transmission of voice calls and other forms of 3 days ago · The registration process from an ATA or IP Phone includes a contact address would be 4042265555@192. type=friend. Mar 4, 2024 · A SIP trunk is merely a collection of virtual phone lines, also known as channels. SIP Responses are the replies you receive. A DTMF tone can be a SIP INFO packet with a specific body to be interpreted by another SIP endpoint. See More ›. [1060] ; This will be WebRTC client. Go to solution. 0+bower. pipe(callSession. 0 without any modification to the source code of SIP. Version 0. localStream; const remoteStream = session. While not intended for all use cases, SimpleUser is intended to be suitable for many single page web browser applications. Some common requests include “Invite” (that’s your call invitation), “Register” (saying, “I’m here!”), and “Bye” (for ending the call). This May 28, 2018 · Importing the library itself is easy enough, but the issues I'm running into are: WebRTC support: instead of using the browser's WebRTC functionality (which isn't present in a react native app), I included react-native-webrtc, and modified SIP. displayName: "Alice ¶€ĸøĸø" hackIpInContact. Adjust Your Voice Settings JsSIP: The JavaScript SIP Library. Oct 16, 2023 · Modifying the call degrades data during transmission or receipt. To report any bugs or suggest improvements head to the GitHub repo. js. It controls the beginning, end, channels, and users during the call. In addition to that, it uses less hardware and resources compared to Asterisk to build powerful apps. com) are assumed to be SIP phones or. The default Session Description Handler included with SIP. refer(target, options). We ported the SIP stack of the p2p-sip project from Python to JavaScript and created an example web-based video phone application for demonstration. The UI is designed to be launched as a popup from within your application. In the land of SIP, the term user agent refers to both end points of a communications session. ctxSip is a Javascript based SIP client that uses WebRTC and WebSockets to connect to your SIP server. The originator of the request creates a locally unique string. ¶. Nov 14, 2014 · host=dynamic. To do this in SIP. Step 2: The call travels through your cloud or premise-based PBX, where it goes out to an IP network. Registration is the first step in making VoIP work. Latest version: 3. Mar 10, 2022 · I'm new to the world of VoIP. User A is located at PBX A. A Messager is required to send the message. 8. Nov 30, 2022 · Now, let’s understand how SIP calling works and its role in VoIP. js is a full-featured SIP stack written in TypeScript. unhold (); hangup() Hangup a call. How SIP calls work. 9, defines a purpose parameter currently with "info", "icon", and "card" tokens. 2. A REFER occurs when persons A and B have an active call session, and A wants to transfer B to speak with C. keepAliveInterval. Basically, a SIP call employs a protocol to allow VOIP calls to connect one device to another. js 0. It is an important part of Internet Telephony and allows you to harness the benefits of VoIP (voice over IP), providing a rich communication experience. audio: true, // only audio calls. Agents (UAs): Alice and Bob. stopSendingKeepAlives() Internal Clears the keepAlive interval from the global timer provider and sets the instance parameter to null. webex. It’s a signaling protocol that manages the initiation, maintenance, and termination of real-time sessions. PBX A is connected to Gateway 1 (SIP Gateway) via a T1/E1. 105. 1) Server work with registration different way - contact server support. Call flow between Gateway-to-Cisco SIP IP Phone Call—Successful Call Setup and Call Hold . 2) You use domain name/realm which is incorrect and domain name/realm is set on server. A mini site that aims to make creating typical SIP flow diagrams easy by using the brilliant mermaid graph library. NO_ACK ‘No ACK’ An incoming INVITE was replied to with a 2XX status code, but no ACK was received. It allows for Overview. A session is nothing but a simple call between two endpoints. 3. If it is an incoming SIP session that has not been established, you need to reject the session. invite() make transfer session. A common mistake is that SIP calling is the only way to use SIP. A SipManager object. These sessions can include voice calls, video conferences, and multimedia distributions across Internet Protocol (IP) networks. 2) You dialling in incorrect format or this number not allowed to call with your extension. 114. EXPIRES ‘Expires’ Incoming call contains an Expires header and the local user did not answer within the time given in the header. com, and it works great. host=dynamic ; Allows any host to register. In SIP there are several ways to end a session depending on what state you are in. Call trunking is your key to unlock the future of business telecommunications. SIP is the basis for VoIP communications, and SIP Trunking is used to provide VoIP connectivity through a PBX. This document defines one new purpose value and one new generic parameter for Call-Info. Once you have instantiated a SIP client, you can access the media streams for a SIP call. Apr 9, 2024 · In short, SIP trunking is used for call setup, management, and teardown. Twilio allows you to set the caller ID for call transfer use cases using the console. Listener. SIP contributes to voice and video calls by May 7, 2024 · Session Initiation Protocol (SIP) is a powerful foundation for modern communication systems. npm install sip. See the Receive a Call guide on how to receive a call. The Messager constructor requires that the User Agent, target URI, and message content be passed into it. When the SIP trunk is enabled over the current internet connection, it allows traffic to bypass PRI lines and connect directly to the PSTN via the internet. Click “+” to add new SIP account. “SIP call flow” is a fancy term to describe how a SIP call works. js, mobile apps, or other platforms, you can define a custom Session Unmute the local media of a call. Android (Native) iOS (Cordova) The SIP. Let’s walk through core API concepts as we tackle some everyday use cases. A SIP call uses Session Initiation Protocol standards to enable Voice over Internet Protocol (VoIP) telephony, transmitting phone calls as data over the Internet. It plays a crucial role in Voice Over Internet Protocol (VoIP) by managing the messages sent between endpoints and handling the actual elements of a call. Also connect to Zoom meeting using ID@zoomcrc. SessionDescriptionHandler represents a common interface for SIP. SIP over WebSocket (use real SIP in your web apps) Audio/video calls ( WebRTC) and instant messaging. Jun 4, 2024 · In basic terms, VoIP is a technology that allows you to make and receive calls over the internet. Oct 4, 2017 · Hi I need to implement something like SIP phone but with a 'classic' SIP without WebRTC. Use A Headset. 168. They’re like your friend saying, “Sure, let’s chat,” or SIP phones connect to servers which initiate contacts with other extensions or route calls. A strong and stable connection is essential for clear and consistent call quality. † The port on which the gateway is prepared to receive the RTP data is specified. With SIP. js has been tested with Asterisk 16. avpf=yes ; Tell Asterisk to use AVPF for this peer. notification event handler. FreeSWITCH. bind(this)) function onUserMediaObtained(stream) { localMediaStream = stream } Simple UI. Contribute to onsip/sipjs-examples development by creating an account on GitHub. When data is lost and is unrecoverable, it’s easy to be disconnected. It replaces traditional analog phone lines, connecting the private branch exchange (PBX) to the internet. co blog post. js library. callSession. I am not using any of the JS libraries for reasons not relevant to this question. A SIP call uses a virtual connection to route traffic from one phone to another, while traditional PRI calls are routed over a physical connection. It works for voice, video, and messaging communications. For a mini blog post about the origin of this project see this falkus. If you want, receiving SIP calls on Android is also possible. SIP may use other transports. Included in the invitation, when setting up a call, are parameters describing exactly what form the audio or video will use. Remove a=imageattr lines from the SDP produced by Jitsi calling a WebRTC endpoint with certain codecs. Mobile Guides. Step 3: Enter the SIP account details provided by your service provider. The target can be either a valid URI or a SIP. SIP offers all potentialities of the common Internet Telephony features like: call or media transfer. 11-29-2018 09:27 AM - edited ‎03-18-2019 02:29 PM. Sep 28, 2012 · Call-ID: 68781700-f791ec0f-2d26-e28690a@10. SIP uses port 5060 both for UDP and TCP. outputAudioStream) // connect local audio file to output stream. Unlike many of the complex protocols in telecommunications, SIP protocol is easy. It can also be anywhere in the 3 days ago · SIP is a signalling protocol used to create, modify, and terminate a multimedia session over the Internet Protocol. Mar 2, 2024 · Concord, CA 94520, This document describes a usage of the SIP Call-Info header field that incorporates Rich Call Data (RCD) associated with the identity of the calling party in order to provide to the called party a description of the caller or details about the reason for the call. 1. 10. An attended transfer occurs when A creates a session with C before connecting B to speak with C. Dec 10, 2023 · SIP, which stands for Session Initiation Protocol, is a signaling protocol primarily used in communication services. on('userMedia', onUserMediaObtained. js you must call sesion. Mar 25, 2021 · Tab more “Three points” icon or settings. Dec 12, 2023 · Secondly, it sets performance and scalability benchmarks with its independent call instance feature. The call is established well, but there's a 40 sec delay between calling the "call" method and establishing a call (starting an RTP session). username=1060 ; The Auth user for SIP. Certainly, several other characteristics can help businesses build more scalable telephony solutions using FreeSWITCH. I've built a client side app in Reactjs that needs to connect with a SIP server to make and receive calls. You can access and act upon calls initiated by a remote caller in a telnyx. On July 11, 2019, the Department submitted a SIP revision to comply with the requirements RFC 3665 SIP Basic Call Flow Examples December 2003 Proxy 1 inserts a Record-Route header into the INVITE message to ensure that it is present in all subsequent message exchanges. It’s used for initiating, maintaining, and terminating communication sessions that include voice, video, and messaging applications. Easy to use and powerful user API. refer(holded_session) Functions hold() and unhold() are not documented in documentation, but when you output session into console, you will see its in there. js and OnSIP — a perfect pairing for WebRTC!. The Call-Info header field, defined in [ RFC3261] Section 20. Jan 21, 2024 · Elevates the flexibility, cost-effectiveness, and simplicity of business telephony. RingCentral: Most comprehensive. FreeSWITCH Legacy. Runs in the browser and Node. Confirm that the SIP URI you have configured for your Trunk's Origination settings is correct. Select SIP settings. Dec 19, 2023 · Here is a detailed step-by-step process for setting up a SIP call on Android. Apr 1, 2024 · SIP. Send a REFER request. An example demo app of SIP. In order to make calls and send messages, create a SIP Simple instance. const content = 'Hello World Getting Started. js Mobile Guides will show you how use SIP. Cause: Twilio is getting no response from your SIP infrastructure. I got past WebRTC support errors, but I don't know if it actually works It must NOT be enclosed between double quotes even if the given name contains multi-byte symbols (SIPjs will always enclose the display_name value between double quotes). This guide assumes that your application is using the built in Session Description Handler in a standard Web Browser with full WebRTC support. js event API. js or Asterisk. The most interesting benefit of SIP is how it cuts down on communication costs. A user picks up the handset of a SIP phone and dials a number to place a call. There are 100 other projects in the npm registry using jssip. Select the Trunk of your choice, in the "General section" under "Call Transfer (SIP Refer)" you can see "Caller ID for Transfer Target". js is fast, lightweight, and easy to use. js is a JavaScript library for WebRTC-based communication. SIP helps in establishing a call by sending signals to each terminal. To make a blind transfer you should provide a SIP URI. Example simple. Lost Connections – You can easily lose the call altogether with this router service. js to work with your softswitch or SIP platform service. Dec 5, 2023 · SIP trunking is a method of delivering telephone and other communication services over the internet using the Session Initiation Protocol (SIP). 3 days ago · Diagram of a request, acceptance, setup and termination of a call. An endpoint can be a smartphone, a laptop, or any device that can receive and send multimedia content over the Internet. The web phone supports audio, video and Internal Sets up an interval to call sendKeepalive() periodically. Simple differs from the full SIP. Also clears the outstanding keepAliveTimeout. SIP Library for JavaScript. With JsSIP any website can get Real Time Communications features using audio, video and more with just a few lines of code. This is called a transfer, and these transfers can be attended or blind. The Session Initiation Protocol ( SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. Some older implementations also add an “@” and its host name to the string. It’s flexible, reliable, and quite lightweight. Once a call is established, other protocols monitor the transfer of audio and data between the two phone systems. Alice (sip:alice@atlanta. 120. js interacts with WebRTC to provide voice, video, and data streams. 15. It has found its MAJOR use in the world of IP Telephony. connection. 120 is the IP address of the endpoint. hold (); unhold() Take a call off hold and unmute the local media. Maybe I should solution is to use software like webrtc2sip? Renegotiation. For instance, the examples on the Demo page are implemented using the SimpleUser class exclusively. js with WebRTC. encryption=yes ; Tell Asterisk to use encryption for this peer. While VoIP can function without SIP, SIP is a versatile protocol that provides compatibility with a Email works over three different protocols: SMTP, IMAP, and POP. The contentDisposition field should be set to render. Apr 28, 2021 · const localMp3FileStream = require('. [2] SIP is a core component of VoIP, the technology that allows you to make and receive calls over the internet. Initiate SIP sessions via the REST API by POST ing to the same calls resource used to initiate traditional phone calls (see making calls for more information). displayName: "Alice ¶€ĸøĸø" hackCleanJitsiSdpImageattr. unmute (); hold() Place a call on hold and mute the local media. Incoming call was not answered in the time given in the configuration no_answer_timeout parameter. With a SIP call, there are more chances to incorrectly modify data packets in unexpected ways, which adversely affects SIP in JavaScript. RCD includes information about the caller beyond the telephone Jul 20, 2022 · SIP trunking is the process of transmitting voice communications using a SIP system. Step 2: Open the app, navigate to settings, and choose “Accounts” or “SIP Accounts. Step 1: The call originates from a SIP Phone — here is where your call is placed, whether you are using a hard phone or softphone. Whether you’re making a call to a SIP address or receiving one, it’s always free with OnSIP. It must NOT be enclosed between double quotes even if the given name contains multi-byte symbols (SIPjs will always enclose the display_name value between double quotes). SIP. † The media capability that User A is ready to receive is specified. Set a random IP address as the host value in the Contact header field and Via sent-by parameter. If registration work ok, can be. If you have changed the FreeSWITCH configuration you may need to update the user details below. This means you can focus on your business instead of the technicalities. Here is an example of how to do this: session. There are libs like JsSIP even with support for WebSockets in Node. com) and Bob (sip:bob@biloxi. Session State Change Full API Demo. We use our SX20's to connect to our WebEx rooms by dialing the WebEx Meeting ID @ourdomain. Below diagram illustrates a successful gateway-to-Cisco SIP IP phone call setup and call hold. Written in TypeScript. Start using jssip in your project by running `npm i jssip`. This section of the documentation is intended to help you configure SIP. sessionDescriptionHandler. Renegotiation allows you to do things such as add video in the middle of a call, put a call on hold, or change codecs that you are using. The call object is created whenever a new call is initiated, either by you or the remote caller. Step 3: The call will travel over to your internet through an NNI Jan 30, 2024 · Accessing Media Streams. SIP message bodies for phone calls are defined in SDP -the session description protocol. You may need a valid SSL Certificate for FreeSWITCH to function properly with WebRTC. Overview. It implements tranaction and transport layers as described in RFC3261. Asterisk Legacy. OnSIP. 0 - uncompressed, minified, bower install sip. After click button and call function callSip our SessionState changes to Ringing, then successful invitation, then SessionState is "Answered" and after that the function onAccept in requestDelegate object is run, but no connection is established because the SessionState goes to "Ended". Much of the client's interaction with the SIP stack happens through May 21, 2020 · Let’s go over the route step by step. Download production and development versions of the SIP. Session Initiation Protocol, or SIP, is one of the protocols for setting up real-time communications over the Web. Aug 29, 2011 · 2. Overview This document provides example call flows detailing a SIP implementation of the following traditional telephony services: Call Hold 3-Way Conference Consultation Hold Find-Me Music on Hold Incoming Call Screening Unattended Transfer Outgoing Call Screening Attended Transfer Call Park Instant Messaging Transfer Call Pickup Unconditional Call . Jan 10, 2018 · When I want to mute microphone I use mediastream which I get from . Sequence Call Flow Generator. With SIP trunking, a phone call is broken down into digital packets and transferred over the internet to a final destination. SIP typically sends these messages in UDP (User Datagram Protocol) on port 5060, with 5061 used for a second line on a two line ATA*(see below). const localStream = session. js#0. js in Node. Mar 8, 2019 · This update to the NOX SIP Call provides states flexibility to establish alternative monitoring and reporting requirements for certain sources not otherwise required to perform Part 75 monitoring under other programs such as the Acid Rain Program or CSAPR. For example, make a SIP call by POST ing to your account's calls list resource URI: Apr 15, 2024 · Google Voice: Best free VoIP service. The example provided will register to FreeSWITCH as user 1000 and will place a call to user 1001. The SIP. Here’s technical overview broken down in six steps. Tired of fighting with configs? Try SIP. 1) You use incorrect secret for this username. 0 401 Unauthorized" message and a nonce to be used to build a hash to be sent in the response. JsSIP is a simple to use JavaScript library which leverages latest developments in SIP and WebRTC to provide a fully featured SIP endpoint in any website. js, you need to create a SIP user agent and register it with a SIP server. The Media Address is where to receive the media or voice (RTP) and could be the same address as the endpoint, 192. Any help on how to connect to the SIP server and how to steam RFC 5359 SIP Service Examples October 2008 1. Proxy 2 also inserts itself into the Record-Route header. The designers of SIP aimed to provide a call setup and signaling protocol End Call. Overview - JS Call. This will be an out of dialog sip message. Web. 120 where 192. Oct 14, 2022 · Before making or receiving a SIP call online, it’s important to check your network connection. Level 1. hold() make new call ua. js in that it will handle attaching media onto the page. The Call-ID header field is an identifier used to keep track of a particular SIP session. This allows you to reference the code for SimpleUser as a reference point for the full SIP. In this way, SIP doesn’t work alone during a VoIP call. Dialpad: Best use of artificial intelligence (AI) Intermedia Unite: Best all-in-one unified communications as a service May 2, 2016 · I am writing a small Node module to make a simple SIP call through Asterisk. on('connected', function() {. This way, individuals and businesses don’t need a legacy telephone line in order to connect. When a call connects, the server and phones negotiate call quality and codecs in real-time. Support early media, hold and transfers. Lightweight! 100% pure JavaScript built from the ground up. statusAssert() Sep 8, 2021 · SIP, or Session Initiation Protocol, is a communication protocol that operates over the Internet. To make an audio call, you should set up a SipAudioCall. The fields needed to send a DTMF INFO are the contentDisposition, contentType, content. A user agent (UA for short) is generally a software agent that is acting on behalf of a user. See full list on sipjs. Utilize SIP in your web application via SIP over WebSocket. The value "jcard" is to be used to associate rich call data related to the identity of the calling party in the Nov 29, 2018 · SIP Dialing Failing. Configurable by configuration. Eliminate the constraints of physical phone lines with a purely digital solution. This destination can be another SIP phone, a VoIP (voice over IP) phone, or a traditional analog phone. You can listen for incoming calls using the SIP. example. Please see the API Documentation to determine how to send an in dialog message. 80. In this scenario, the two end users are User A and User B. The default value is Transferee and you can change to Transferor based on your use cases. Use your VoIPstudio SIP username, Password and Domain details to complete the SIP profile under the android device. js, you can harness the power of WebRTC to build audio, video, and realtime data into your application. The ACK (F15) and BYE (F18) both have a Route header. The fact that SIP is an open standard has sparked enormous interest in the telephony market, and In SIP to make a transfer you must send a REFER message to the endpoint that you have a session with. js tries to leave the majority of handling media to the user application. [1] SIP is used in Internet telephony, in private IP telephone systems, as well as mobile phone calling over LTE ( VoLTE ). sms("Sms text") In other words, i just need to receive a nodeReadableStream from the SIP network with the sound of the voice that the person speaks to the SessionDescriptionHandler represents a common interface for SIP. Extends the functionality of legacy phone systems while improving reliability and quality. Call Proceeding—Gateway 1 to PBX A Gateway 1 sends a Call Proceeding message to PBX A to Free SIP calls with OnSIP. It helps two or more parties have a successful, productive interaction. secret=1060 ; The SIP Password for SIP. To handle incoming calls with SIP. Send DTMF RFC 2833 or SIP INFO. This tells the endpoint to render the content Feb 26, 2019 · saved current session in other variable, for example var holded_session = session; call in current session hold, session. Calls within your OnSIP network of users, which are Jan 3, 2024 · To make an audio call, you must have the following in place: A SipProfile that is making the call (the "local profile"), and a valid SIP address to receive the call (the "peer profile"). This guide will go over starting an audio only call and then adding video to it. js web apps. // send sms. Send instant messages and view presence. brandonthor. 1, last published: 7 months ago. You will need to create the body of the packet to send. As of SIP. Helpful links: Example applications using SIP. 9. The UA also maintains the WebSocket, on Nov 16, 2023 · SIP Requests are like writing a letter to someone, asking them to call you. Easiest way is npm. When I send the INVITE message, as expected, it returns with a "SIP/2. Check your firewall to be sure the Twilio IP addresses and ports are allowed. js API. ”. hangup (); sendDTMF(tone) Send a DTMF event via SIP Message to the active call Set CallerID for call transfer. com with no issues. js to interact with the underlying RTP connection. Once the call is connected, Twilio will then fetch the TwiML you specify for the call. js') localMp3FileStream. js, mobile apps, or other platforms, you can define a custom Session Nov 26, 2020 · When I make a call I successfully invite destination URI. Configure Asterisk. js Server Configuration Guides will show you how to configure softswitches to work with SIP. We will use the UserAgent. 0 renegotiation is supported through the reinvite() and hold() functions. SIP is a text-based protocol that uses UTF-8 encoding. A Call is the representation of an audio or video call between two browsers, SIP clients or phone numbers. SIP Session Establishment. We do not use anything outside of the API to create the SimpleUser. js session. Then go to Advanced settings. sip. Note that Chrome and Firefox on Android are WebRTC-capable and compatible with SIP. SIP-enabled devices. js maintains the SimpleUser interface which is a wrapper around our full API. SIP, on the other hand, is a protocol through which many VoIP calls are facilitated. All that is fine. js can be used to build voice and video communication applications in React Native. /localAudioFileStream. x has introduced a new API (currently in beta), with new documentation autogenerated from our source. Get started now. See the Make a Call guide on how to make a call. js associates a SIP address to a UA, and that SIP address can make and receive requests on that user’s behalf. When using SIP. Calling the SIP. Works with OverSIP, Kamailio, Asterisk, OfficeSIP and more ( more info) Jan 2, 2023 · SIP is just one method of deploying VoIP; its primary benefit is the fact that it provides a direct connection between private or local telephone systems and the public telephone network. Simple() method, with options will create a new Simple object. js the application needs to be aware of the state of the session and call the proper method to end the session. If you’re dialing a number with a SIP address associated with it, OnSIP will automatically recognize it and handle the call, meaning that it will also be free. The SimpleUser class provides an easy simplified interface for making audio and video calls in a web page. Step 1: Go to the App Store and download a SIP app like Krispcall. RFC 3665 SIP Basic Call Flow Examples December 2003 3. This will send a message to a targeted URI. js on mobile platforms. This project provides a complete SIP stack in JavaScript for implementing SIP based audio and video user agents in the browser or mobile. A headset reduces background noise and improves call quality, and it’s crucial in a noisy environment. Create real-time peer-to-peer audio and video sessions via WebRTC. Asterisk. az nq at et us cl eh gp hw eu