White Paper Conquering Scalable WebRTC Conferencing

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1 Conquering Scalable WebRTC Conferencing

2 Executive Summary Developers are embracing WebRTC technology for building their next generation services but leveraging only peer-to-peer topologies may not be enough. Instead having a deeper understanding of optimizing media processing through a variety of topologies is required for scalability, reliability and advanced functionality. These types of features can only be achieved with server-side media processing. But questions of which topology is best, when one topology vs another should be implement, and the drawbacks of each topology can often lead to confusion. To help answer these questions, the white paper will discuss: How peer-to-peer differs from peer-to-server and what advanced features are available when introducing a centralized media server How using a multi-point control unit (MCU) topology can benefit mobile and limited bandwidth clients How using a selective forwarding unit (SFU) topology can benefit performance and user-interface flexibility The benefits of having both topologies in a converged architecture available on-demand 2

3 Table of Contents An Introduction to WebRTC Infrastructure Peer-to-Peer Versus Peer-to-Server... 4 Peer-to-Peer Mesh Mess... 5 Multipoint Control Unit (MCU)... 6 Selective Forwarding Unit (SFU)... 7 Converged Architectures - Applications For the Real World... 8 Next Generation Hybrid-SFU/MCU... 8 Conclusion About Dialogic

4 An Introduction to WebRTC Infrastructure Since its release by Google in mid-2011, the WebRTC (Web Real-time Communications) project has been a disruptive technology, converting the internet browser into a powerful multimedia engine. WebRTC brings together state-of-the-art real-time communications technologies, incorporating advanced audio and video codecs (Opus and /9), mandatory encryption (SRTP & DTLS) and Network Address Translation traversal (ICE & STUN). As originally defined, WebRTC was specified as a peer-to-peer technology and since its inception, WebRTC has drastically reduced the barrier to entry for web developers to build real-time communication applications through simple JavaScript APIs. But to be clear, WebRTC is a technology, not a finished application or service. While WebRTC was originally envisioned as a purely peer-to-peer technology, many everyday business applications require centralized media functions, improving reliability, efficiency, or scaling through a peer-to-server architecture. A deeper understanding of the issues and trade-offs between peer-to-peer and peer-to-server architectures are important in building WebRTC applications. Peer-to-Peer Versus Peer-to-Server WebRTC was designed to send media directly between clients via their browsers, also known as peer-to-peer (P2P). In the peer-to-peer architecture, communication between the clients is preceded by first establishing a signaling connection to the application server (sometimes referred to as a signaling server). The signaling method or protocol is not specified within the WebRTC specifications, thus allowing the adoption of an existing method (SIP, WebSockets, XMPP, etc.) or implementing a proprietary signaling process. The application server holds the business logic and acts as the intermediary for the Session Description Protocol (SDP) exchange. Once the SDP exchange completes, direct media communication between the two clients can begin. Application server Alice Signaling Signaling Bob Media Figure 1 Peer-to-peer While WebRTC is designed to be primarily browser-to-browser, there are a growing number of use cases that benefit significantly when media is anchored in the network with a server to act as a media peer, also known as peer-to-server (P2S). Similar to peer-to-peer, in the peer-to-server architecture, the clients again establish a signaling connection to the application server. In this architecture, the application server continues to manage the business logic but also utilizes a media control connection to the server for the SDP exchange between the client and the media server. Once the SDP exchange completes, media communication between the client and server can begin. Application server Alice Signaling Media control Signaling Bob Media Media Media server Figure 2 Peer-to-server Utilizing server-side processing can introduce advanced functionalities such as centralized recording for compliance purposes, audio/video playback, media analysis for speech-to-text detection, transcoding for connecting disparate networks, and media mixing for multiparty conferencing. Depending on the architecture, server-side processing can optimize bandwidth and minimize client compute, benefiting mobile clients by increasing battery life and providing a flexible user interface to clients. 4

5 For more complex applications that support multiple participants (conferencing, distance learning, contact center), there are three primary network topologies when it comes to WebRTC: Peer-to-Peer Mesh Multi-point Control Unit (MCU) Selective Forwarding Unit (SFU) Peer-to-Peer Mesh Mess The peer-to-peer mesh topology operates without a centralized media server requiring each client to simultaneously send its encoded media to each participant client in the conference. Synchronously, the client must also receive and decode each participant s media stream. Peer-to-peer mesh is often times referred to as peer-to-peer mess given the amount of streams required. For instance, in a four-person videoconference, each client browser is encoding and transmitting three media streams while also receiving and decoding three additional media streams. VP9 Figure 3 peer-to-peer mesh topology The WebRTC client has full control over the video layout. Media latency is typically not an issue when using peer-to-peer mesh since media will most often be direct to the receiving client. The peer-to-peer mesh appeals to front-end developers because of the low cost to implement. However, these topology architectures are extremely limited in functionality and scale. For instance, without a centralized media server, the client is left to perform advanced features such as recording. Putting this type of functionality at the WebRTC client level not only causes added processing by the client but can also risk compliance requirements. Regarding scalability, the process of encoding/decoding media streams is a compute intensive process - notably encoding being a ~4x more intensive process than decoding. Assuming each client in the peer-to-peer mesh is using the same codec, frame-rate and resolution, the transmitting client could leverage only encoding once. However, clients within the conference often do not share the same codec, frame-rate, and resolution. Therefore, additional encoding is required. Furthermore, bandwidth can be a preventive factor for scale with most notably the potential limited uplink bandwidth from most devices. For example, the bitrate for a single 720p resolution video stream starts at ~1.5 Mbps. In a standard four client conference, the uplink bandwidth required will be ~4.5 Mbps. It is for these reasons, WebRTC clients especially (but not limited to) mobile and tablet devices where compute and bandwidth are limited, will see a reduced scale. In summary, while peer-to-peer mesh topologies produce a flexible user interface and low latency media connections, the disadvantages of scalability, transcoding, and lack of advanced functionally are a significant deterrent to launching many applications. Server Client Advanced Functionality Conference size = N Peer-to-peer mesh NA NA High (N:N) High (N:N) Video layout Client Latency Record/ playback Transcoding Lowest Client-side No 5

6 Multipoint Control Unit (MCU) The use of Multiple Control Unit (MCU) topologies have been popular for real-time communications well before the inception of WebRTC. MCU topologies operate with a centralized media server to which WebRTC clients send their encoded media stream. From there, the MCU media server receives the video stream, decodes the stream, tiles the decoded frames with the streams from other participants, and then encodes the tiled video to send it back to the participant. Using the MCU topology architecture simplifies the stream the WebRTC client will need to send and receive and reduces the number of streams to just one. H.264 H.264 Figure 4 Multiple Control Unit (MCU) topology SIP Reducing the required encode/decode to one will therefore decrease both the client compute and bandwidth consumption, thus benefiting mobile type devices. Furthermore, since each stream is decoded and transcoded at the MCU media server there is no concern for each WebRTC client to share the same codec, frame-rate, and resolution profile. The incoming media streams from the various clients can now be transcoded to another codec, trans-sized to a different resolution, and trans-rated to a different frame rate thereby allowing each client to be optimized to their preferred profile. For instance, a mobile client preferring to utilize H.264 with VGA resolution and 15 frames-per-second can be connected to a laptop WebRTC client using the codec with 720p resolution and 30 frames-per-second. Furthermore, since the MCU media server is acting as a peer, quality of service (QoS) can be applied on a per stream basis thus not allowing one poor connection to dictate the quality of all users of a multi-party application. A key benefit of the MCU topology is that it shifts the processing of the encode/decode from the client into the server, often as part of a cloud compute service, where processing resources are less expensive. The performance issue is becoming more important as newer video coding systems are very computationally intensive and conference users expect a high quality, high resolution video such as 720P and 1080P at 30 frames per second or higher. As with peer-to-peer topology, sharing encoders across multiple client streams can address the performance drawback and result in greater server scalability. And lastly, the MCU will utilize the same tiled video composition for each connecting client. By doing so, the video layout in a multiparty conference will be dedicated by the server-side for all client participants. In summary, multipoint control unit topology results in a significantly lower compute and bandwidth for the clients and can interconnect disparate networks through transcoding/trans-rating/trans-sizing with the disadvantages of around server performance and an inflexible client user interface. However, the centralized computational resource of an MCU can become a limiting factor in cost-sensitive, large-scale, one-to-many applications. Server Client Advanced Functionality Conference size = N Video layout Latency Record/ playback Transcoding Multipoint Control Unit (MCU) High* (N:N) High (N:N) Normal (1:1) Normal (1:1) Server Acceptable Server-side Yes * Can be reduced with encoder sharing 6

7 Selective Forwarding Unit (SFU) Selective Forwarding Unit (SFU), also known as video routing, is a topology allowing for WebRTC clients to send their encoded video stream to the centralized media server where it is then forwarded/routed to the other WebRTC clients. The SFU topology is an attractive approach to addressing the server performance issue, as it doesn t involve the compute expense of video decoding and encoding. Additionally, without encoding/decoding, the latency of the added SFU media server is minimal. Lastly, the clients with full correspondence with the SFU media server have complete control over the streams it receives, and because the client is receiving the streams it wants, it can have full control over the user interface flexibility. VP9 Figure 5 Selective Forwarding Unit (SFU) topology While the SFU topology has become a popular choice among WebRTC communities, perhaps the most common overlooked shortcoming of SFU topology is the default to using the least common codec. This means every participant in the conference need to use the same codec. For those multiparty conferences where all participants are using PC/laptops, the issue is negligible but introducing a mobile device that is hardware optimized for H.264 acceleration would be better suited using a different codec. The inability to transcode video streams can limit the type of clients that can be connected together. Furthermore, the multiple streams being routed/forwarded by the SFU to the WebRTC client can cause increase in downlink bandwidth, thus causing increased decode processing. The issue can be mitigated by limiting the number of streams being forwarded to the client to either the active talker, a subset of the streams, or a combination of both. Additionally, using a method called simulcast allows multiple streams to be encoded by the WebRTC client. Typically, two streams with the first being encoded with high resolution and second encoded with lower resolution. This way, the SFU can forward/route the high definition stream of the active talker while still sending the lower definition streams of the listeners. Lastly, traditional SIP based platforms cannot handle the multiple streams produced by the SFU. For this reason, without a gateway function in the middle, the SFU topologies are restricted to WebRTC only. In summary, selective forwarding unit topology produces a flexible client user interface and improved server-side performance; the disadvantages of requiring a least common codec and client compute/bandwidth are concerns to be considered. Server** Client** Advanced Functionality Conference size = N CPU Encoders: Decoders Uplink: Downlink CPU Encoders: Decoders Uplink: Downlink Video Layout Latency Record/ Playback Transcoding Selective Forwarding Unit (SFU) Moderate (1:N) Moderate (1:N) Medium (1:N) Medium (1:N) Client Low Server-side No ** Will be more with simulcast 7

8 Converged Architectures - Applications For the Real World As discussed, no topology is perfect, and each come with distinct advantages as well as disadvantages. Multipoint control unit architectures are ideal for when compute and bandwidth are limited and there is a need for interoperability with disparate networks but come at a cost of high server load and limiting video layout. On the other hand, selective forwarding unit topologies are ideal for high server performance and maximum flexibility for the client UI but come at a cost of requiring all connecting clients to share the same codec, frame-rate, and resolution profile. The tough decision is which to use for your application. Launching an application into real world scenarios requires on-demand access to both the capabilities of a MCU and the capabilities of a SFU. This complementing feature set has paved the way for a new, next generation hybrid-sfu/mcu architecture. Server Client Advanced Functionality Conference size = N Video layout Latency Record/ playback Transcoding Peer-to-peer mesh NA NA High (N:N) High (N:N) Client Lowest Client-side No Multipoint Control Unit (MCU) High (N:N) High (N:N) Normal (1:1) Normal (1:1) Server Acceptable Server-side Yes Selective Forwarding Unit (SFU) Moderate (1:N) Moderate (1:N) Medium (1:N) Medium (1:N) Client Low Server-side No Next Generation Hybrid-SFU/MCU The hybrid-sfu/mcu topology allows for the media stream to be delivered based on the preference optimized for the individual client. For example, in cases where the client is a mobile or SIP device the media server can deliver a single MCU-type mixed stream. For WebRTC clients capable of handling multiple streams and no restrictions on bandwidth or compute, then the media server can deliver forwarded/routed-type streams. Additionally, having the ability to transcode individual streams while leaving all others to be forwarded/routed eliminates the least common codec issue of SFU. Allowing these capabilities to be utilized from within the same media server can give the application server s business logic the most flexibility in delivering a scalable, reliable, feature-rich service. VP9 VP9 H.264 SIP Figure 6 hybrid-sfu/mcu topology 8

9 Conclusion Real-time communications is forever evolving and the benefit of having server-side infrastructure has been realized for many years. WebRTC has certainly paved the way for advanced next generation services to be deployed but understanding the benefits infrastructure can bring to the service is essential for scale and reliability. About Dialogic Dialogic is a leading cloud-optimized solutions provider for real-time communications media, applications, and infrastructure to service providers, enterprises, and developers around the globe. Based in Parsippany, NJ with offices worldwide, Dialogic helps 48 of the world s top 50 mobile operators, and nearly 3,000 application developers build and deploy on agile networks. To learn more about Dialogic s WebRTC solutions portfolio and product roadmap, contact your local Dialogic representative or visit us on the web at 9

10 For a list of Dialogic locations and offices, please visit: INFORMATION IN THIS DOCUMENT IS PROVIDED IN CONNECTION WITH PRODUCTS OF DIALOGIC CORPORATION AND ITS AFFILIATES OR SUBSIDIARIES ( DIALOGIC ). NO LICENSE, EXPRESS OR IMPLIED, BY ESTOPPEL OR OTHERWISE, TO ANY INTELLECTUAL PROPERTY RIGHTS IS GRANTED BY THIS DOCUMENT. EXCEPT AS PROVIDED IN A SIGNED AGREEMENT BETWEEN YOU AND DIALOGIC, DIALOGIC ASSUMES NO LIABILITY WHATSOEVER, AND DIALOGIC DISCLAIMS ANY EXPRESS OR IMPLIED WARRANTY, RELATING TO SALE AND/OR USE OF DIALOGIC PRODUCTS INCLUDING LIABILITY OR WARRANTIES RELATING TO FITNESS FOR A PARTICULAR PURPOSE, MERCHANTABILITY, OR INFRINGEMENT OF ANY INTELLECTUAL PROPERTY RIGHT OF A THIRD PARTY. Dialogic products are not intended for use in certain safety-affecting situations. Please see for more details. Dialogic may make changes to specifications, product descriptions, and plans at any time, without notice. Dialogic is a registered trademark of Dialogic Corporation and its affiliates or subsidiaries. Dialogic s trademarks may be used publicly only with permission from Dialogic. Such permission may only be granted by Dialogic s legal department at 3300 Boulevard de la Côte-Vertu, Suite 112, Montreal, Quebec, CANADA H4R 1P8. Any authorized use of Dialogic s trademarks will be subject to full respect of the trademark guidelines published by Dialogic from time to time and any use of Dialogic s trademarks requires proper acknowledgement. The names of actual companies and products mentioned herein are the trademarks of their respective owners. Dialogic encourages all users of its products to procure all necessary intellectual property licenses required to implement their concepts or applications, which licenses may vary from country to country. Any use case(s) shown and/or described herein represent one or more examples of the various ways, scenarios or environments in which Dialogic products can be used. Such use case(s) are non-limiting and do not represent recommendations of Dialogic as to whether or how to use Dialogic products. Copyright 2016 Dialogic Corporation. All rights reserved. 12/

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