What is the difference between WebRTC, XMPP, and SIP?

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WebRTC, XMPP, and SIP are all technologies used for real-time communication, but they serve different purposes and operate in different ways. Here’s a breakdown of each:

WebRTC (Web Real-Time Communication)

Purpose: WebRTC is designed for enabling real-time audio, video, and data sharing between browsers and mobile applications without the need for plugins.
Key Features:
Peer-to-peer communication: Allows direct connections between users.
Supports media streaming: Handles video and audio codecs for high-quality communication.
Data channels: Enables the transfer of arbitrary data between peers.
Use Ca

WebRTC, XMPP, and SIP are all technologies used for real-time communication, but they serve different purposes and operate in different ways. Here’s a breakdown of each:

WebRTC (Web Real-Time Communication)

Purpose: WebRTC is designed for enabling real-time audio, video, and data sharing between browsers and mobile applications without the need for plugins.
Key Features:
Peer-to-peer communication: Allows direct connections between users.
Supports media streaming: Handles video and audio codecs for high-quality communication.
Data channels: Enables the transfer of arbitrary data between peers.
Use Cases: Video conferencing, online gaming, file sharing, and customer support applications.

XMPP (Extensible Messaging and Presence Protocol)

Purpose: XMPP is primarily a messaging protocol for instant messaging and presence information.
Key Features:
Decentralized architecture: Users can connect to different servers and still communicate.
Extensible: Supports various extensions (XEPs) for features like file transfer and multi-user chat.
Presence information: Allows users to see the availability status of their contacts.
Use Cases: Chat applications, social networking, and IoT messaging.

SIP (Session Initiation Protocol)

Purpose: SIP is a signaling protocol used for initiating, maintaining, and terminating real-time sessions in VoIP (Voice over Internet Protocol) applications.
Key Features:
Call setup and teardown: Manages the establishment and termination of calls.
Can handle multimedia sessions: Supports voice, video, and messaging.
Works with various transport protocols: Can operate over UDP, TCP, and TLS.
Use Cases: VoIP services, video conferencing, and unified communications.

Summary

WebRTC is focused on peer-to-peer communication directly in web browsers, enabling real-time media and data sharing.
XMPP specializes in messaging and presence, providing a decentralized framework for chat applications.
SIP is geared towards managing and controlling sessions for voice and video calls, primarily in VoIP environments.

Each technology has its strengths and is often used in conjunction with others to create comprehensive communication solutions.

Dave Cridland

Expert in XMPP, implemented 3 servers. · Author has 548 answers and 3.3M answer views

· 7y

Let’s start with WebRTC. WebRTC gives you a bunch of nebulous magic that eventually emits some Session Description Protocol (SDP) lines which describe a media session you’d like to bring up. If you exchange this with the other end somehow, then you’ll hopefully end up with the session live and transmitting.

What WebRTC doesn’t give you is any way of exchanging the SDP with anyone else. It doesn’t have any idea what “anyone else” even means. You have a few options - loosely, you can write your own signalling infrastructure, or else use one that already exists. An advantage of using a standardized protocol is that someone else - and usually many someone elses - have carefully designed and reviewed it already to ensure that it works efficiently and securely. Another is that you’ll be able to get existing (often open source) server-side infrastructure.

SIP is an existing, standardized, signalling protocol. It’s already used by softphones, and some hardware phones as well. It deals with SDP directly, and figuring out how to get it to another SIP endpoint. This is useful, but in practical terms, SIP has a few limitations, and one of these is that it’s close to impossible to figure out if the other endpoint will be able to connect back. Another is that SIP is suprisingly tricky to embed in a web browser.

XMPP, on the other hand, is rather more versatile. It is also a standardized protocol used for instant messaging, but it can also handle arbitrary applications quite easily. It has a signalling protocol called Jingle which is (still) used by Google for Hangouts, Meet, etc.

Jingle’s downside is that it’s not native SDP - but there’s plenty of existing Javascript code for translating.

XMPP’s big upside is that it can operate over plain TCP, but also over Websockets and tunnelled through HTTP quite easily. These are all standardized and so you’ve a wide choice of servers to deploy.

So, a summary: You need WebRTC and a signalling protocol. Your signalling protocol can be any of XMPP, SIP, or something you write yourself.

What is Genesys SIP (session initiation protocol) server development?

Why is XMPP not more popular in web development?

What are the advantages of WebRTC over SKYPE? Why technology giants rushing to adopt webRTC?

Which is the most high-performance XMPP server?

Why doesn't Skype use SIP and/or open source its protocol?

Jason Wills

Product Specialist at vCloudx Pte Ltd

· 4y

Session Initiation Protocol (SIP) is a signalling protocol used in VoIP technology. It is used for initiating, maintain, modifying and terminating real-time multimedia communications sessions between endpoints over the IP networks.

Extensible Messaging and Presence Protocol (XMPP) is an open XML technology for real-time communication. It is an open communication protocol designed for instant messaging(IM) and chats. It is ideally suited for point-to-point and conference chat session.

Web Real Time Communications (WebRTC) is a free, open project that provides browsers and mobile applications with

Web Real Time Communications (WebRTC) is a free, open project that provides browsers and mobile applications with real-time communications capabilities. In WebRTC the choice of signalling protocol is left open to the developer. He can choose either SIP, XMPP or any other custom signalling protocol for doing the SDP exchange and peer discovery.

In short, SIP is ideal for implementing a full-featured telephone system while XMPP is ideal for implementing a full IM platform and, via extensions like Jingle, allows you to initiate calls and other media sessions. WebRTC is ideal for browser/app-based video/audio communications instead of developing your own proprietary stacks.

Arun P

Head - Unified Messaging Solution

· 6y

WebRTC is a free, open project that provides browsers and mobile applications with real-time communications capabilities.

XMPP framework for establishing p2p sessions makes for a great pairing with WebRTC.

XMPP is particularly a great fit with WebRTC in settings where there is a desire to pair WebRTC audio/video calls with text chat, but the advantages of XMPP

Because WebRTC is a peer-to-peer protocol, multi-user experiences become exponentially complex. Pairing a WebRTC service with XMPP allows developers to dramatically reduce this complexity.

SIP used for flexible and scalable connectivity of V

WebRTC is a free, open project that provides browsers and mobile applications with real-time communications capabilities.

XMPP framework for establishing p2p sessions makes for a great pairing with WebRTC.

XMPP is particularly a great fit with WebRTC in settings where there is a desire to pair WebRTC audio/video calls with text chat, but the advantages of XMPP

Because WebRTC is a peer-to-peer protocol, multi-user experiences become exponentially complex. Pairing a WebRTC service with XMPP allows developers to dramatically reduce this complexity.

SIP used for flexible and scalable connectivity of VoIP infrastructure.

PC: Mirrorfly

Andrew Zhilenko

CTO of PortaOne, vendor of VoIP/billing solution for 400+ telcos worldwide

· 2y

Between WebRTC, XMPP, and SIP, which is better for creating a video chat application for a mobile device?

First of all, the question needs clarification - now it sounds a bit like “What should I use to make Sangria - apples or oranges?” (for those unaware - Sangria is wine, mixed with fruits and served with ice, very popular in Spain during summer). Just like you actually need both apples and oranges for Sangria - you need both some type of signalling protocol (which would be SIP) and media transport (which would be WebRTC or RTP) to build a video-chat app.
Bu if we look for “optimal technology stack to build video chat app” - then (in 2023) WebRTC is definitely your optimal choice for the media part. You can use the library already embedded into major browsers or mobile OS to save tons of time on dealing with network communications or media decoding. You will have to pick a signaling protocol though. You can go with SIP (and there are libraries like SIPjs that would allow you to use it from your app), but you will ...

Can we use XMPP for developing a video streaming (for video call through web application) application?

Have you used Twilio for building WebRTC video conferencing into the site?

Which is better MQTT or XMPP for group chat application?

In VoIP, what is SIP trunking/SIP trunks? What are the advantages?

How do you scale XMPP to multiple servers?

Andrew Zhilenko

CTO of PortaOne, vendor of VoIP/billing solution for 400+ telcos worldwide

· 2y

What is the difference between SIP and WebRTC technology?

SIP (Session Initiation Protocol) is a signaling protocol used for initiating, maintaining, modifying and terminating real-time sessions that involve video, voice, messaging and other communications applications and services between two or more endpoints on IP networks. In other words in controls when to start a call, when to end, etc. - it does not actually know anything about the actual conversation contents, which is sent as a media stream using RTP (Real-Time Transport Protocol) between end-points. RTP was designed for carrier networks (i.e. servers sitting on public IP addresses with no f

WebRTC (Web Real-Time Communication) is a free, open-source project that provides web browsers and mobile applications with real-time communication (RTC) via simple APIs. It allows audio and video communication to work inside web pages by allowing direct peer-to-peer communication, eliminating the need to install plugins or download native apps. WebRTC is supported by most modern web browsers and is used by many applications, including video conferencing, file transfer, and voice over IP (VoIP). WebRTC itself does not know (and does not control) the signalling (start/stop of a call), so it has to be implemented separately using some other API. WebRTC cannot do magic and bypass a firewall, but it has a lot of “standard” tools that can be used to facilitate a connection between two users behind NAT of home router or firewall.

So while a SIP and WebRTC are both used for real-time communication over the internet, but they have some differences:

SIP is a protocol for initiating and terminating communication sessions, while WebRTC is a set of APIs already embedded in web browsers and some OS to initiate and receive real-time audio and video stream directly. So WebRTC is more comparable with RTP with a few differences:

Imagine you are a developer who wants to create an application like Skype or Zoom. You have two possible approaches:

implement in your application handling of RTP streams, and then add capabilities of dealig with network communications, receiving & decoding the data - and then playing audio stream and displaying the video. Tedious work - plus it requires a lot of experience in the areas where a typical front-end developer never steps (and does not really want to step ;-) ).
Use WebRTC library already embedded in the browser and let it handle the complex issues of networking, media conversion, etc. Embed a WebRTC widget into your app and add some basic signaling - and voila! you have everything working.

In summary, SIP is a signaling protocol used to set up and tear down communication sessions, while WebRTC is a set of APIs and libariies that enables real-time audio and video communication in web browsers and mobile applications.

Dave Cridland

Expert in XMPP, implemented 3 servers. · Author has 548 answers and 3.3M answer views

· 10y

What's the difference between SMPP and XMPP?

One letter.

If you want a more serious answer, it gets tricky. It's not tricky because they're so similar, it's tricky because they're so different.

SMPP is a protocol for sending SMS and pager messages - I first ran into it building a notification system for an ISP I worked at. It's a pretty straightforward protocol, and it actually has more in common with protocols like X.400 - it's a single PDU per transaction - but the methodology is closer to SMTP - it has a very constrained view of the universe, and has extensibility bolted on afterward.

Fundamentally, it remains a protocol for sending a te

One letter.

If you want a more serious answer, it gets tricky. It's not tricky because they're so similar, it's tricky because they're so different.

Fundamentally, it remains a protocol for sending a text message. It has little idea of character encoding, so you're restricted into hoping the SMSC you're talking to have the same character set you're expecting. Addresses are typically raw telephone numbers - but not always - and there's no routing protocol, just peer-to-peer. This is fine if the intent is "I'm going to connect to my service provider and send a bunch of SMSs", but that's as far as it goes.

XMPP, on the other hand, is a fully routed protocol, with scoped addressing (so we have domains to route with), arbitrary payloads, a defined character encoding (UTF-8), and so on. It's not restricted to messages, you can do query/response interactions too, and you can broadcast presence state around the network.

You can, also, bridge from XMPP into SMPP pretty easily, for simple messages - but the XMPP endpoint that's mapped to an SMPP endpoint will be capable of vastly less rich interaction than a native one. In return, you'll have lost the distinct error codes of SMPP.

You can consider this like a feature chart:

SMPP Endpoint Features:

1) Receives textual messages.
2) Returns SMSC specific errors.

XMPP Endpoint Features:

1) Sends/Receives textual messages.
2) Sends/Receives arbitrary XML messages.
3) Asks/Answers arbitrary XML queries.
4) Broadcasts XML state.

With those primitives, even more complex systems can be (and have been) built. My desktop client will watch for geolocation updates from my friends. It'll tell me what tune they're listening to. It'll show me a picture of them. I can join chat-based meetings, have a voice call - I can also see who's available for a voice call. I can even send someone a text message.

We could map SMPP to XMPP fully, so you could locate a SMSC over XMPP, and have a lossless translation of error codes and other protocol features. It's not wholly clear you'd want to; if you need all that stuff, and you know your SMSC address, just use it.

Really, comparing SMPP and XMPP won't get you very far, because you're implicitly thinking that they're fundamentally the same kind of protocol - and they're simply not.

Nikolay Sapunov

CEO at ForaSoft. We develop video chatting applications

· Updated 3y

What is the difference between Push Notifications, WebSockets, WebRTC?

For starters, I would say these are completely separate definitions. However, all of them can be used in real-time communication apps. The following image illustrates how these work in a video chatting app on mobile.

Push notifications - this is a one-way style of communication when the user subscribes to receiving updates/ messages. The important feature of Push notification is that it delivers messages to the client when the application is not running or the website is not opened. From a technical perspective, it works like Websocket with the difference that there is an intermediary in a form

Push notifications - this is a one-way style of communication when the user subscribes to receiving updates/ messages. The important feature of Push notification is that it delivers messages to the client when the application is not running or the website is not opened. From a technical perspective, it works like Websocket with the difference that there is an intermediary in a form of an operational system notification server, such as APNs for Apple and
WebSockets - is a communication protocol that allows bidirectional interaction between a client-side (an app or browser) and the server in real-time. WebSockets only work when the application is running or the web page is opened. Websockets are not designed for media-rich data transfer.
WebRTC really stands out from the list. It’s a browser-oriented real-time communication standard. This means a transfer of large amounts of media data with close to zero latency between client devices. Moreover, WebRTC is usually about peer-to-peer connections, meaning that the data flows directly between users devices. Although, to start a WebRTC session users need to exchange information about the session first (like when to start a session, who to connect with and what codecs should be used). This means that WebRTC has no signaling of its own and this is necessary in order to open a WebRTC peer connection. This is achieved by using other transport protocols such as HTTPS or secure WebSockets. In that regard, WebSockets are widely used in WebRTC applications.

To sum up, these are completely different concepts in the world of development. All three of them provide different functionality and can be used all in one application.

Hope my answer helps!

John Romkey

Software Design and Development, Security, IoT · Author has 210 answers and 766.8K answer views

· 6y

What is the best one for IoT (Internet of Things) Technology -- SIP (Session Initiation Protocol), XMPP (Extensible Messaging and Presence Protocol ), or REST (Representational state transfer)?

Here’s the thing about “IoT” - the Internet of Things isn’t all one thing.

What’s the particular application or device you’re trying to build?

Different applications and devices have different needs.

Some need to send lots of data (video, audio,), some need to send a little once in a while (door sensors).

Some can tolerate lots of latency (delay) in the network (most sensors), some can’t (live video).

Some have rich interactions with lots of possible types of data, some are just going to send small variations on the same thing over and over again (most sensors).

Trying to use one technology across t

Here’s the thing about “IoT” - the Internet of Things isn’t all one thing.

What’s the particular application or device you’re trying to build?

Different applications and devices have different needs.

Some need to send lots of data (video, audio,), some need to send a little once in a while (door sensors).

Some can tolerate lots of latency (delay) in the network (most sensors), some can’t (live video).

Some have rich interactions with lots of possible types of data, some are just going to send small variations on the same thing over and over again (most sensors).

Trying to use one technology across the entirely of IoT applications

In particular to your question:

SIP would be a peculiar choice as it’s generally used in telephony and signaling applications. I’m not sure why you’d want to use it for, say, a sensor.

XMPP would also be a peculiar choice as it’s generally used for chat and real-time communication among people. You could layer sensor data (for instance) over it - there are specifications for doing this, even specifically for IoT, but frankly it’s a damned weird application for it. XMPP is a heavy protocol for such a simple thing. And remember that many CPUs used in IoT applications are small and slow, without a lot of resources - not well suited to heavy protocols.

REST is much more appropriate tech, best suited for applications moving modest amounts of richer data. You wouldn’t use it for video streams but you might use it to set them up. Be aware that proper REST is much more restrictive than just “build an API over HTTPS”; true REST methodology dictates that the actions the API provides map strictly onto HTTP methods and manipulate objects in fairly specific ways. REST is not a good fit for all API-based applications.

MQTT is often used in IOT applications - it’s lightweight, fast and efficient. It’s great for doing things like sensors. You wouldn’t want to use it for, say, authenticating and authorizing users, or video streams.

You could stand on your head and jump through hoops and force almost any protocol into service for almost any application. You could take sensor readings, run them through a voice synthesizer, use SIP to place a VOIP call to someone who’d listen to the reading and record it in a database. It’d be absurd and an insanely bad use of resources but you could do it.

So if you’re dead set on using whatever random protocol, you can probably make it work. You just shouldn’t.

You’re much better off seeking the minimal solution for the particular problem you’re trying to solve, or device or application you’re trying to build. Use the simplest protocols and the simplest data representations (spoiler alert: in 2018 that’s rarely XML). You’ll need fewer resources in the devices, less developer and devops time, and you’ll have a smaller attack surface so you’ll at least have a chance at being somewhat secure, unlike a huge amount of the IoT space.

Tsahi Levent-Levi

Vast experience in VoIP and video calling. Independent consultant for VoIP and WebRTC. · Author has 282 answers and 853.3K answer views

· 9y

Which is the better protocol for instant messaging and in future maybe VoIP (Voice over Internet Protocol) and video chatting, XMPP or WebSocket?

Comparing WebSocket to XMPP is impossible.

Split the technologies into two parts:

Signaling protocol
Transport protocol

WebSocket is a transport protocol. One on which you can use signaling of your own. XMPP can be used on top of WebSocket for example (see RC 7395 - An Extensible Messaging and Presence Protocol (XMPP) Subprotocol for WebSocket).

Does XMPP fits best as your signaling protocol for instant messaging, would it be SIP SIMPLE (or god forbid RCS) or should you use your own proprietary solution? That would depend on the use case you have in mind and to what type of service are you trying t

Comparing WebSocket to XMPP is impossible.

Split the technologies into two parts:

Signaling protocol
Transport protocol

Arjun Kava

Working on webRTC since 2015

· 3y

How does WebRTC work?

WebRTC is a Torrent of video and audio communication. It uses on peer to peer protocol to provide communication between two devices.

It is not that simple as I described in the first line but in layman terms, it is bypassing the client-server model which enables direct communication between two peers in the network.

However, there are many challanges while bypassing server such as getting address of peer behind NAT(Network Adress Translation) and Firewall, sync both peers while establishing and persisting communication (I know too technical).

One of the major advantage of webRTC is, it is plug an

WebRTC is a Torrent of video and audio communication. It uses on peer to peer protocol to provide communication between two devices.

It is not that simple as I described in the first line but in layman terms, it is bypassing the client-server model which enables direct communication between two peers in the network.

One of the major advantage of webRTC is, it is plug and play framework with support of all modern browsers and native devices such as Android, IOS etc.

I’ve keep it simple to understand in this answer althouhg if you want to explore more technical details read following answer:

Original Question: How does WebRTC work?

Candra Ramsi

What are the pros and cons of XMPP (Extensible Messaging and Presence Protocol) versus web sockets for real-time web communication?

XMPP protocol could be used through web socket. So i am assuming you meant json through web socket. Since both are using websocket then i am comparing json message passing vs xmpp message passing. I would say the pros of using xmpp are

Interoperability between client version
Security (?)

I am not very experienced in xmpp but i would say that it has been very well thought over the years. When building a new message passing system you should think about if the older solution is adequete rather than wasting building new system with the same capability.

The cons of xmpp would be xml size and the numbe

Interoperability between client version
Security (?)

The cons of xmpp would be xml size and the number of presence would a nightmare after a while. Message archiving would also a bit of a problem but you could turn it off if you don't need message history.

If you only need message passing to server then i would not suggest xmpp. If you want to build a chat system the i would recommend going from any open sourced xmpp server and client then modified them to suit your needs.

Priologic Victoria

EasyRTC - full-stack WebRTC open source bundle

· 10y

What is the quality of existing WebRTC options when it comes to latency on VOIP, Jitter, and the load resilience of a WebRTC VOIP solution?

I think before you start testing you have to really think carefully about what use case you're testing for.

Is it just audio or both audio and video? Is there data being exchanged? Are you doing a mobile native application, too? What Operating system and CPUs are you targeting? Are you using just Chrome or do you want Firefox and Opera browsers too? Are you going to use a SaaS offering or host your own. Are you going to go more towards a centralized architecture or are you going more peer-to-peer? Are there bandwidth constraints?

It's really complicated and the best test might be to develop

I think before you start testing you have to really think carefully about what use case you're testing for.

It's really complicated and the best test might be to develop a proof of concept.

What I do know is that the Google code base is getting lots of attention. So it should continue to improve. I'm not sure any single vendor can compete directly with that in the long run. While it is open source it is really big. Read millions of lines of code and those lines of code are changing as we speak.

So it will take the industry some time to absorb it. We've been working on it for nearly two years and we're getting a good understanding of it.

There are some knobs in the WebRTC native source that you can adjust to tune WebRTC for your particular application and set of circumstances. We're playing in that area now, specifically for native mobile apps and getting some reasonable results.

Ashish Sharma

B.S. in Master of Science in Computer Science & Software and Applications, University of California, Berkeley (Graduated 2011) · Author has 159 answers and 97.5K answer views

· 5y

How does WebRTC work?

WebRTC enables high-quality audio and video communication within the web browsers. Web real-time communication is a powerful and robust technology and increasingly being adopted by business solutions. It is an open-source framework, developers by Google. WebRTC is supported by all the prime web browsers. it is a platform-independent technology and does not need any additional plugin to set up the communication across the devices.
You can understand the webRTC application development with the below-mentioned components

Get User media: It gets access to the audio and video present in the device.

RTC peer communication: This components of webrtc enable audio and video communication by using signal processing, codec handling, and bandwidth management.

RTC Data channel 1: It possesses a reduced latency and uses API for enabling communication between the two users.

Getstats: It is used to decipher the statistics of the webrtc sessions in the devices. WebRTC uses HTML5 and simple APIs for peer to peer communication. It is a progressive technology and undergoing development process for more sophistication.

Jason Wills

Product Specialist at vCloudx Pte Ltd

· 4y

What is the difference between Push Notifications, WebSockets, WebRTC?

Push notification is a clickable pop-up message that appears in users' mobile phone. It is a delivery of information to the recipients from an application server where the transaction is initiated by the brand. Recipients do not have to be in the app or using their devices to receive them. This is different from in-app notification where the message only appears when the user is within the app

WebSocket is a technology that enables a two-way interactive communication session between the user's browser and a server. The bi-directional, full-duplex messages mechanism allows simultaneous transmission of data in both directions through a single connection. This is unlike HTTP which provides half-duplex communication.

WebRTC (Web Real-Time Communication) enables web applications to capture and stream voice and video media, as well as to exchange arbitrary data between browsers without requiring an intermediary. When starting a WebRTC session, you need to negotiate the capabilities for the session and the connection itself. This can be done either via HTTP or WebSocket.

Albert Abello Lozano

Former WebRTC researcher. · Author has 101 answers and 225.3K answer views

· 9y

What is WebRTC and what can it do?

Has been a while since I researched about WebRTC but I think I can answer this question.

WebRTC is an Application Programming Interface (API) developed by the W3C that is still on development and being standardized by the RTCWEB IETF group (Rtcweb Status Pages).

Its main goal is to allow peer-to-peer communication between web browsers. This communication can be used to transfer multiple type of data: files, video or audio. All this without the need of any plugin.

Of course this type of project arises many problems involving different areas, security, protocols, multimedia codecs etc. However, it

Has been a while since I researched about WebRTC but I think I can answer this question.

WebRTC is an Application Programming Interface (API) developed by the W3C that is still on development and being standardized by the RTCWEB IETF group (Rtcweb Status Pages).

Of course this type of project arises many problems involving different areas, security, protocols, multimedia codecs etc. However, it is in a pretty advanced status and it can be used right now: AppRTC.

Right now Chrome and Firefox are two of the browsers that offer full WebRTC support on PC.

Technically the API is divided into three different parts:

getUserMedia
PeerConnection
DataChannels

getUserMedia is a JavaScript library used to access the media inputs of the device through the browser (microphone and/or webcam). Once the inputs are allowed the PeerConnection library starts a dialing to connect to the other peer and agree on the data transfer method. DataChannels allow both browsers to share the data obtained previously.

The discussion groups have been working hard to bring this to life going through multiple problems but seems it is going to be releasing its full commercial potential soon as many startups and corporations are starting to use this API for their software.

WebRTC is part of HTML5 .

Aleksandr Blekh

Consultant, researcher, SW engineer, data scientist, educator, entrepreneur · Author has 497 answers and 1.4M answer views

· 10y

What are the chances (and approximate time frame) of seeing seamless interoperability between Skype and SIP-based VoIP systems, including the plugin-free browser-based ones (WebRTC)?

So far Microsoft has been enjoying Skype's enormous user base. However, by abandoning its system interoperability features (Web presence indicator, SDK, APIs, gateways, etc.), the company alienated many developers and some users. I understand that development ecosystem is not nearly as important to Skype's business model, as to other Microsoft's product lines, but ignoring the convenience factor (promised by WebRTC) might result in many users leaving Skype for more open VoIP platforms.

It seems to me that Microsoft/Skype's management starts realizing the potential danger and risks, associated w

It seems to me that Microsoft/Skype's management starts realizing the potential danger and risks, associated with it. Hence the recent PR, focused on presenting company as a supporter of open inter-connectivity solutions. On the other hand, as Tsahi Levent-Levi mentioned in his answer, they recognize the risk of advancing too fast and wide on this front, which would jeopardize Skype's monetization due to potential migration of some users to other platforms. Hence, Microsoft's efforts appear to proclaim interoperability goals in general, while, in practice, they are more likely to be focused on this as a PR campaign as well as improving their own product ecosystem (Lync, Internet Explorer, etc.).

Therefore, it looks like we should not expect Microsoft's widespread and significant efforts on achieving VoIP systems interoperability between Skype and open standards VoIP platforms (unless they will see significant enough migration of users).

Manisha Bharti

User of WordPess ,Blogger · Author has 325 answers and 492.2K answer views

· 5y

What does WebRTC mean?

WebRTC stands for web real-time communications. It is a very exciting, powerful, and highly disruptive cutting-edge technology and standard. WebRTC leverages a set of plugin-free APIs that can be used in both desktop and mobile browsers, and is progressively becoming supported by all major modern browser vendors. Previously, external plugins were required in order to achieve similar functionality as is offered by WebRTC.

WebRTC leverages multiple standards and protocols, most of which will be discussed in this article. These include data streams, STUN/TURN servers, signaling, JSEP, ICE, SIP, SD

WebRTC leverages multiple standards and protocols, most of which will be discussed in this article. These include data streams, STUN/TURN servers, signaling, JSEP, ICE, SIP, SDP, NAT, UDP/TCP, network sockets, and more.

Abul Walid

Mobile and VOIP Specialist

· 7y

What is the difference between Asterisk and WebRTC for real communication?

Hi,

Asterisk is a software pbx (class 5 switch) which is a central part of any ip telephony system.

On the other hand webRTC is a technology that enables real time media into a browser. webRTC can be used to built a voip client that connects to asterisk and make a phone call, or can be used to built a conference client that connects to a conference mixer.

webRTC does not have any signalling protocol, it is only media layer and network layer needed for media, media setup is done using SDP (session descriptor protocol).

SIP(session initiation protocol) can be added separately to implement signalling

Hi,

Asterisk is a software pbx (class 5 switch) which is a central part of any ip telephony system.

webRTC does not have any signalling protocol, it is only media layer and network layer needed for media, media setup is done using SDP (session descriptor protocol).

SIP(session initiation protocol) can be added separately to implement signalling which is used by asterisk for signalling. Based on the need anything can be used for signalling like custom connection over web socket.

Neha Dhamija

Works at Voiphorizon

· 8y

How UDP make SIP (Session Initiation Protocol) reliable?

Hello,

UDP is used for delivery of services such as Voice, Video.
These services create user experience and if there is a drop in packet then the only option is to proceed without waiting for a re-transmission.

That being said, if there is a drop in packet, then concept of jitter buffers come into picture. As per the SIP RFC

Hello,

That being said, if there is a drop in packet, then concept of jitter buffers come into picture. As per the SIP RFC

“The INVITE transaction consists of a three-way handshake. The client transaction sends an INVITE, the server transaction sends responses, and the client transaction sends an ACK. For unreliable transports (such as UDP), the client transaction retransmits requests at an interval that starts at T1 seconds and doubles after every retransmission. T1 is an estimate of the round-trip time (RTT), and it defaults to 500 ms. Nearly all of the transaction timers described here scale with T1, and changing T1 adjusts their values.”

You might want to refer to the following link for more details,
SIP: Session Initiation Protocol

Thanks,

Neha Dhamija

[1]

Footnotes

[1] VOIP - Horizon

Enkode Technologies

What is the difference between WebRTC and WebSockets?

Web Sockets are use for a server per session and designed by bi-directional communication between client and server. On the other hand WebRTC is relating to networks in which each computer can act as a server for the others.

David Knell

I want to develop a Voip service that includes video. The service should work from mobile devices. Should it be built based on SIP, WebRTC or something else?

Probably not "something else." Can't really answer the rest without knowing more about the application. Your server end will most likely be built on SIP.

Then clients - WebRTC in the browser isn't a comprehensive answer (at least, not when I last looked) - iOS doesn't support it. SIP clients are two a penny, but do you want to support a bunch of different clients for different devices? WebRTC for those that support it and apps for those that don't looks like a reasonable client strategy at this point.

Skip Shekhani

BS in Industrial Engineering (college major) & Psychology, Iowa State University (Graduated 1985) · Author has 23.8K answers and 6.2M answer views

· 2y

What are the differences between WebRTC and WebEx?

Webex - Connect with anyone, anywhere, any time. WebRTC - A free, open project that provides browsers and mobile applications with Real-Time Communications.

Buğra Çakır

Protoman

· 13y

Why did Google choose XMPP instead of SIP as VoIP protocol for Gtalk?

I think Jingle (http://en.wikipedia.org/wiki/Jingle_(protocol)) has some major improvements and advantages over SIP. One of them is lower effort for software development with XMPP-Jingle and i can also mention that Jingle has teleconferencing support built-in the protocol itself.

David Johnson

10+ years of exp in Voip

· 5y

What is the difference between Asterisk and WebRTC for real communication?

Asterisk Service and Solution

Asterisk is an open source, converged telephony platform, which is designed primarily to run on Linux. The power of Asterisk lies in its customizable nature, complemented by unmatched standards-compliance. No other PBX can be deployed in so many creative ways as Asterisk!

Scott Helms

Working in telecommunications since 1998. · Author has 2.8K answers and 7.3M answer views

· 5y

What is the difference between the SIP and RTP protocols used with VOIP?

SIP provides presence and call control. RTP contains the actual audio. Most people are only aware of SIP in terms of voice, but it can be used in many different communication scenarios including chat and video conferencing.

Shrikant Umarye

General Manager · Author has 9.7K answers and 15.1M answer views

· 5y

What is the difference between the SIP and RTP protocols used with VOIP?

Sending Voice over IP (VoIP) requires two protocols: SIP and RTP (Real-time Transport Protocol). SIP – Protocol used for establishing a session (call). IP address and port information are exchanged. RTP – After SIP establishes a session, this protocol is used for exchanging voice packets.

Jeremy Martin

CTO @ Stream

· 9y

What is the functionality of WebRTC?

Here's an excerpt from the MDN documentation:

WebRTC (Web Real-Time Communications) is a technology which enables Web applications and sites to capture and optionally stream audio and/or video media, as well as to exchange arbitrary data between browsers without requiring an intermediary. The set of standards that comprises WebRTC makes it possible to share data and perform teleconferencing peer-to-peer, without requiring that the user install plug-ins or any other third-party software.

WebRTC consists of several interrelated APIs and protocols which work together to achieve this. The documentat

Here's an excerpt from the MDN documentation:

WebRTC (Web Real-Time Communications) is a technology which enables Web applications and sites to capture and optionally stream audio and/or video media, as well as to exchange arbitrary data between browsers without requiring an intermediary. The set of standards that comprises WebRTC makes it possible to share data and perform teleconferencing peer-to-peer, without requiring that the user install plug-ins or any other third-party software.

WebRTC consists of several interrelated APIs and protocols which work together to achieve this. The documentation you'll find here will help you understand the fundamentals of WebRTC, how to set up and use both data and media connections, and more.

In simpler terms, WebRTC is a collection of Web API's that facilitate real-time communication between web-connected devices. It can be used to create apps similar to Hangouts (without the need for browser plugins), but importantly, it also exposes lower-level API's (such as RTCPeerConnection) that expose domain-agnostic data-sharing utilities, which could be used for, say, file transfers, screen sharing, or internet radio.

Nicolas Vérité

Former member of the XSF's Board of the Directors

· 9y

Which is the better protocol for instant messaging and in future maybe VoIP (Voice over Internet Protocol) and video chatting, XMPP or WebSocket?

XMPP's Jingle is much simpler to build and maintain than SIP, but less interoperable with operators/carriers. It is quite similar, and still based in ICE (STUN+TURN) and UDP.

XMPP brings also chat and PubSub... Which are huge advantages for social media and much more.

Najeeb Shaikh

Programmer & Writer · Author has 70 answers and 269.1K answer views

· Updated 8y

What are possible implementations of XMPP Protocol?

A2A.

It really depends on your requirements. Over the years XMPP has been extended from being just a simple user messaging protocol to becoming many things for many people. Remember that the protocol has been designed such that it really doesn’t care about the nature of its payload. A few use-cases where the XMPP protocol could be, or is already being used.

Messaging. This is the plain vanilla case of the protocol, of course. But then messaging itself as a domain has evolved tremendously over the years. Think how you could add your own layer of functionality, a differentiator possibly, that will

A2A.

Messaging. This is the plain vanilla case of the protocol, of course. But then messaging itself as a domain has evolved tremendously over the years. Think how you could add your own layer of functionality, a differentiator possibly, that will make your XMPP-enabled product stand out. I would recommend that you have a look at both Flock and Slack. These two products have very nicely evolved the whole idea of chat as a basic communication primitive into a making it a core business function. In fact, both products seek to become viable alternatives to email to a very large extent (not entirely, though).
Presence & Notification. Remember that XMPP does not just have messaging in it, but also presence. Presence is nothing but information about the availability of a user as s/he becomes available or unavailable. But then, presence need not be restricted solely to humans. How about devices? How about using it to power smart homes? It could also be used for interacting with remote systems. The possibilities are endless.
Group Messaging. This is another area that is quite distinct from one-on-one user chats. Group messaging is another major growth area for messaging especially for collaboration, something very similar to IRC.
VOIP. Google extended the protocol with its jingle extension, so that the same protocol can now be used for voice chats.
Location-based Systems. As mentioned earlier, XMPP does not bother about the payload. So how about carrying geolocation data of a mobile user, perhaps? Or even for tracking mobile assets like trucks and inventory?
Business Workflows. There is an extension for this too, so that you can design entire workflows around XMPP. The extension as a standard supports all well-known business actions required for management and execution of a workflow, like interactions between two business users, or even between a user and an automated business sub-process, and so on.

I have listed only a few of the use cases for XMPP off the top of my head. I am sure other answers here will highlight even more.

HTH.

Vijay Parvatikar

Product Leader, Technology Expert, Evangelist - Online Video · Author has 71 answers and 259.7K answer views

· 3y

Which one is better, CMAF or WebRTC?

As of early 2022, I can surely say WebRTC is the lowest-latency real-time communication protocol, which has ultra-low latency with sub-300 ms video delivery and can scale up to millions of viewers. Platforms built on WebRTC can provide us meet all our requirements of latency, scale, reach, quality and finally the cost.

Safari 11 supporting Webrtc has only made it better to support all iOS devices.

While CMAF (HTTP based) is very relevant, popular and is applicable in several use-cases and is making video streaming much cheaper and less complicated but fails to provide the lowest latency like WebRTC. Latency may not be your primary goal in your use-case as compared to quality (or less buffering) so in those cases,...

Dave Cridland

Expert in XMPP, implemented 3 servers. · Author has 548 answers and 3.3M answer views

· 5y

Is the XMPP service provider able to access the conversation content of users?

There are an astonishing number of end-to-end encryption (or E2EE) systems available for XMPP. Most of these are ephemeral encryption technologies, but all of them, in principle, will make it impossible for a server operator to read the conversations passing through it.

Ephemeral encryption - using a different key for each message that is then discarded - mean that reading messages from data collected at the server even if a key is recovered from the device is impossible, too.

However, ephemeral encryption also forces the device to maintain a local, unencrypted, archive of the messages - so if y

However, ephemeral encryption also forces the device to maintain a local, unencrypted, archive of the messages - so if you can obtain the key from the device, you may as well just get that instead. As ever, then, the whole question around E2EE is “What is your threat model?”

So far, this is the same for any messaging system.

Where XMPP differs from other messaging systems is in two ways.

Firstly, most conversations across one XMPP server probably pass through two, and group conversations pass through three. This means that the number of service providers is higher than the likes of WhatsApp, but also means that network analysis - who talks to whom - is impossible across the entire network - protecting you against an entirely different attack.

Secondly, and much more importantly, in order to obtain the public keys of other participants in any messaging system, one must obtain them from the server. At best, one can then verify them via some other communications channel, such as a QR code. If the server and clients are all built by the same organization, however, none of this proves anything. A malicious service provider (or one simply compelled by a state) can simply inject other keys and lie about the QR code verification. Fundamentally, then, in order to trust the encryption is safe, you need to trust the service provider you use, in which case why worry about the question of E2EE at all?

In the XMPP world, however, anyone can write a client or a server, and many service providers exist, and you can combine all of these however you prefer. Moreover, this is an actual practical reality - there are dozens of different clients and servers which haven’t a single line of code shared between them. This makes an attack based around what we might call the “active security theatre” described above much more difficult to arrange.

Overall, then, I would say that if your threat model is solely concerned with your service provider reading your messages, this is vastly simpler to defend against in XMPP than other systems.

That all said, I remain one of many people in the XMPP community who actively disable E2EE, since the increased risks of server interception are, we feel, outweighed by the decreased risk in device security and the improved user experience a server-side archive and so on provide.

Abdul Basit

Full Stack Web Developer at Upwork (2014–present)

· 7y

What is WebRTC?

I am currently making a webrtc app for a client. So basically what it does it provides peer to peer communication. You can check AppRTC which is the official website where you can test its functionality.

You can make voice call, video call, screen sharing, file sharing web apps which can be direct peer to peer, or can involve a server/gateway for extended functionality like recording on server etc. Example would be kreunto media server, janus web gateway etc.

The best thing about it is that it allows peer to peer communication. I was testing the app I made and was doing a voice call and screensh

The best thing about it is that it allows peer to peer communication. I was testing the app I made and was doing a voice call and screensharing session with my client. After our call had setup, the server crashed due to a bug but the call didn’t disconnect because it was between me and my client directly.

But keep in mind that you need a signaling service to go along with it. You can think of it as you want to talk to your neighbor while you are standing out on your door. You send a child to your neighbors house to ask them to come outside etc. and once they come outside you guys start talking to each other directly. You can think of the child as the signaling service. Ignore the bad example :D

If you need help, let me know I’ll guide you further and can also give you relevant links.

P.S Safari browser doesn’t support webrtc yet. So you’ll have to make a native ios app for webrtc as it’ll not run in the browser. You’ll have to use webrtc native libraries for that.

Tsahi Levent-Levi

Vast experience in VoIP and video calling. Independent consultant for VoIP and WebRTC. · Author has 282 answers and 853.3K answer views

· 10y

What are the chances (and approximate time frame) of seeing seamless interoperability between Skype and SIP-based VoIP systems, including the plugin-free browser-based ones (WebRTC)?

Very little chances for that to happen.

The problem isn't technology but rather business. Today's OTT players have no incentive to open up their networks to others. This would devalue their service and with hurt their ability to monetize their users base.

It can only happen when the OTT in question feels threatened by other OTTs and is trying to find ways to entice users to stay or onboard his platform - something that doesn't usually happen.

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