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Running a retail business is hard enough. Managing your phone system shouldn’t make it harder. If you’ve ever struggled to answer every call, connect staff across stores, or reach customers quickly, you’re not alone. Most communication systems weren’t built with retail in mind. That’s why finding the right retail UCaaS provider matters more than ever. […]
The post What to Look for in a Retail UCaaS Phone System: 7 Must-Have Features appeared first on Sangoma Technologies.
Healthcare practices today face growing communication challenges. Staff are stretched thin, patient expectations are rising, and many organizations are still relying on outdated phone systems—tools that were never designed for the speed and complexity of modern healthcare delivery. In many cases, these systems require staff to manually check voicemails, return missed calls, and track down […]
The post Exploring the Impact of UC on Healthcare Workflow Efficiency appeared first on Sangoma Technologies.
Unified Communications as a Service (UCaaS) brings voice, video, messaging, and collaboration into one cloud-based platform. It’s designed for how modern teams work—whether they’re remote, hybrid, or in-office. With easy deployment, low IT overhead, and built-in scalability, UCaaS helps businesses stay productive and connected without managing servers or juggling apps. What is UCaaS (Cloud UC)? […]
The post Cloud UCaaS Deployment: Simple, Scalable Communications for Growing Businesses appeared first on Sangoma Technologies.
Most clinics don’t need to be told their communication tools are broken—they live it every day. A patient leaves a voicemail that gets buried. A nurse calls back but misses the window. The front desk is juggling calls while trying to check in a walk-in. No one has time to track who said what, let […]
The post How Unified Communications Helps Healthcare Teams Deliver Better Patient Care appeared first on Sangoma Technologies.
In hospitality, every minute matters. When communication breaks down, it shows fast. A request is missed, a handoff fails, a guest ends up waiting—and they don’t forget. It’s rarely a people problem. Most teams are doing their best with tools that were never meant to work together. Desk phones, radios, personal devices, random apps. Everyone’s […]
The post How Unified Communications Improve Customer Satisfaction in Hospitality appeared first on Sangoma Technologies.
For industries with strict compliance requirements or data sovereignty concerns, keeping communication infrastructure in-house can offer added peace of mind. On-Premises Unified Communications (UC) gives you full administrative control over updates, integrations, and data—without sacrificing the collaboration features teams rely on. While cloud and hybrid deployments are secure and scalable (and often the right fit […]
The post On-premises UC: A Reliable Choice for Regulated Industries appeared first on Sangoma Technologies.
If you’ve ever juggled before, you’ve probably let the balls drop. Now, imagine trying to juggle communication between multiple hotel properties. It’s easy to see how things can get missed, calls can get lost, and guests can end up frustrated. Managing multiple properties means managing a lot of moving parts. But one part that often […]
The post Why Multi-Property Hotel Chains Need a Centralized Telephony System appeared first on Sangoma Technologies.
Unified Communications (UC) usually pushes businesses towards a tough choice: the robust control and security of on-premises systems, or the agility and scalability provided by cloud-based solutions. How about if you didn’t have to choose? Hybrid UC delivers precise control over critical infrastructure while harnessing cloud flexibility to scale effortlessly. In this article, we’ll explore […]
The post Why Hybrid UC is the Best of Both Worlds for Scalability and Control appeared first on Sangoma Technologies.
OpenAI is utilizing WebRTC for its Realtime API! Even better, webrtcHacks friend and Pion founder Sean DuBois helped to develop it and agreed to a Q&A about the implementation. It is not often a massive WebRTC use case like this emerges so rapidly. In addition, Sean was extremely transparent about his work at OpenAI. In […]
The post OpenAI & WebRTC Q&A with Sean DuBois appeared first on webrtcHacks.
Healthcare communication should be easy. But too often, it’s a struggle. Outdated phone systems create frustration, confusion, and delays. Patients wait on hold, important messages get lost, and staff spend too much time tracking down information. This isn’t just inconvenient—it’s costly, both for practices and patients. The good news? There’s a better way. By upgrading […]
The post Outdated Phone Systems Are Hurting Healthcare. Here’s How Unified Communications Can Fix It appeared first on Sangoma Technologies.
If your hotel brand can’t deliver a consistent guest experience across every property, you’re losing trust—and revenue. Guests expect the same level of service whether they’re checking in at your flagship resort in Miami or your boutique property in Denver. But for most hotel groups, delivering that kind of consistency isn’t easy. Disconnected communication tools, […]
The post How Hotels Can Use Unified Communications to Deliver High-Quality Guest Experiences Across Every Brand Location appeared first on Sangoma Technologies.
Let’s face it: launching a UCaaS (Unified Communications as a Service) system from scratch is daunting. This type of cloud phone system incorporates many different types of communication/collaboration services to help your business succeed, including voice, video, messaging, meetings, presence, and team-based collaboration. And while the hallmark of a UCaaS environment is its ability to bring […]
The post Easy UCaaS Migration and Deployment: Best Practices and Challenges appeared first on Sangoma Technologies.
Technology has changed how we work—and fast. Teams are spread across offices, homes, and time zones. Customers expect fast responses at any time of day. Meanwhile, businesses are juggling a mess of apps, devices, and phone systems that don’t talk to each other, slow everyone down, and cost too much to maintain. That’s why more […]
The post What is UCaaS? A Complete Guide to Unified Communications as a Service appeared first on Sangoma Technologies.
Thursday, April 10 at 17:00 UTC / 11:00 AM EDT / 8:00 AM PDT OpenAI is utilizing WebRTC for its Realtime API! Join Chad Hart, Editor of webrtcHacks, for an analysis of WebRTC trends in GitHub, OpenAI is utilizing WebRTC for its Realtime API! Join webrtcHacks editor Chad Hart for a Q&A with OpenAI’s WebRTC […]
The post REMINDER: Livestream this Thursday April 10: Open AI WebRTC Q&A with Sean DuBois appeared first on webrtcHacks.
Healthcare providers are under more pressure than ever. Doctors and healthcare staff spend long hours, dealing with mountains of paperwork, managing telemedicine appointments, and answering patient calls. It’s a lot to handle—and it takes time away from what matters most: caring for patients. That’s where Unified Communications (UC) and Generative AI (GenAI) come in to […]
The post Why Healthcare Providers Are Turning to UC with GenAI for Better Patient Care appeared first on Sangoma Technologies.
As Chad mentioned in his post last week, we have been diving into what OpenAI is doing with WebRTC. Over the last months, we actually did a full teardown and compared OpenAI’s Realtime API to what powers chatgpt.com. What intrigued us most was how to measure response latency. One of the key metrics for any […]
The post Measuring the response latency of OpenAIs WebRTC-based Realtime API appeared first on webrtcHacks.
In hospitality, the pressure to provide exceptional guest experiences is always high. Whether you run a boutique hotel or a large resort, guests expect fast, personalized service, and any hiccup in communication can quickly lead to frustration and bad reviews. How can you keep things running smoothly and ensure each guest leaves happy? The answer […]
The post The Role of Unified Communications in Enhancing Guest Experience in Resorts and Hotels appeared first on Sangoma Technologies.
Every missed call, delayed message, or clunky video conference costs your business more than time—it costs opportunities. The average employee switches between 13 different apps 30 times a day, losing focus and wasting hours in the process. Over the course of a week, that’s four hours spent just trying to reorient after toggling between tools. […]
The post The Ultimate Guide to Unified Communications Solutions appeared first on Sangoma Technologies.
Imagine being a doctor at a busy practice. Your office is packed with patients, your team is juggling phone calls, prescription refills, appointment scheduling, and in-person visits—all at once. The phone rings again, but this time, it’s not answered, or worse, it’s dropped entirely. You’re already buried in paperwork, and the outdated phone systems aren’t […]
The post Doctors Losing Patients Over Missed Calls: How Sangoma’s Healthcare Communications Can Help appeared first on Sangoma Technologies.
With big players like NEC pulling the plug on their on-premises offerings and others cutting support for traditional systems, it’s easy to think the future of Unified Communications (UC) lies entirely in the cloud. However, not all companies are on board with this shift. At Sangoma, we’ve committed to continuing to invest in Switchvox On-Premises […]
The post Switchvox Admin Training Videos for Beginners and Experts appeared first on Sangoma Technologies.
OpenAI using WebRTC in its Realtime API is obviously exciting to us here at webrtcHacks. Fippo and I were doing some blackboxing of this based on a quick sample the day of the WebRTC announcement so we could look at it in webrtc-internals and Wireshark. Some weeks later, my daughter was interested in using ChatGPT […]
The post The Unofficial Guide to OpenAI Realtime WebRTC API appeared first on webrtcHacks.
International Women’s Day is a time to celebrate the incredible contributions of women worldwide and reflect on the progress made toward gender equality. At Sangoma, we proudly recognize the women who inspire us daily, driving innovation, growth, and success across our company and the broader business world. As a company rooted in values like trust, […]
The post Celebrating Women in Business: How Sangoma Supports Empowerment and Innovation appeared first on Sangoma Technologies.
The communications world is buzzing with big changes, and it’s leaving many small and medium-sized businesses (SMBs) in a tough spot. Avaya, a long-standing player in the space, has announced that starting June 30, 2025, it will no longer support businesses with fewer than 200 seats on its Avaya Experience Platform (AXP). If you’re one […]
The post Avaya Abandons SMBs – Why Sangoma is the Ideal Solution for Small and Medium-Sized Businesses appeared first on Sangoma Technologies.
We’re excited to share that Jitsi has been accepted into Google Summer of Code 2025! This is a great opportunity to get hands-on experience, collaborate with our awesome community, and contribute to the Jitsi open-source ecosystem.
If you’re interested in participating, check out our idea tracker for some inspiration. If you have your own cool project idea, we’d love to hear it!
To get started, make sure to check out the following resources:
The deadline for applications is April 8th, 2025, so be sure to get your proposal in before then! We’re here to support you throughout the process, and we can’t wait to see what you’ll bring to the table.
Stay tuned for updates, join the community discussions, and let’s make GSoC 2025 amazing!
The post GSoC 2025, let’s do this! appeared first on Jitsi.
I have been updating a WebRTC in Open Source dataset derived from GitHub event data for 10 years now. I periodically update this to look for recent trends on WebRTC activity, popular repos, and new API usage. I hosted a live stream of my 2024 review back in December where Tsahi Levent-Levi joined to help […]
The post 2024 WebRTC in Open Source Review: A Quantitative Analysis appeared first on webrtcHacks.
Debugging WebRTC media issues, especially video, often requires access to the unencrypted RTP payloads. We talked about this back in 2017 already and had a great blog post on using the libWebRTC “video_replay” tool. While that post has aged remarkably well, video_replay has improved significantly, in particular since it is now possible to create the […]
The post Capture & Replay WebRTC video streams for debugging – video_replay 2025 update appeared first on webrtcHacks.
Understanding the FCC’s New STIR/SHAKEN Rules: A Compliance Guide for Businesses
The post The New FCC STIR/SHAKEN Rules and Why They Matter for Your Business in 2025 appeared first on Sangoma Technologies.
The AV1 codec represents the cutting edge of video compression technology. Developed by the Alliance for Open Media (AOMedia), a consortium including tech giants like Google, Microsoft, Netflix, and Mozilla, AV1 is a loyalty-free video codec that was designed to meet the ever-growing demand for high-quality video streaming while minimizing bandwidth usage.
It offers up to 30-50% better compression compared to its predecessors like VP9, VP8 and H.264, enabling sharper visuals at lower bitrates. This makes it an ideal choice for bandwidth-constrained environments, such as mobile networks or rural areas with limited internet speeds. Despite its impressive capabilities, the adoption of AV1 has been slower than anticipated due to several factors. AV1 has higher computational demands as its advanced compression algorithms require significantly more processing power for encoding and decoding. Hardware acceleration for AV1 is still emerging and therefore use of AV1 can result in higher CPU/energy consumption and suboptimal performance on low end devices and those without hardware support.
The path to adding AV1 support in Jitsi was not straightforward. Before we could enable AV1, it was essential to integrate support for the various modes and complexities that the codec offers, both in the Jitsi Meet client and the Jitsi Video Bridge (JVB). Jitsi had to extend JVB’s capabilities to handle AV1 streams, including managing simulcast and SVC modes seamlessly for multi-user conferences. This groundwork laid the foundation for AV1’s eventual inclusion as the preferred codec in Jitsi deployments.
AV1’s RTP encapsulation is unusual, if not weird, compared to RTP payloads for other video codecs – all the information an RTP Selective Forwarding Unit (SFU) like the JVB needs is carried in a “Dependency Descriptor” RTP header extension, rather than in the RTP payload proper. This means that the JVB doesn’t technically need to support the AV1 codec at all – it only needs to support the dependency descriptor header extension.
This format is unusual in that it was developed not in the IETF, where RTP payload formats are normally defined, but rather by AOMedia itself. The main consequence of this is that the header is encoded very much like a video codec: it’s very parsimonious of bit usage, at the cost of being both annoyingly complicated to parse, and being stateful – information needed to parse the header is sent only at intervals. For more information about this complexity, see Lorenzo’s post from several years ago – https://www.meetecho.com/blog/av1-svc/.
Handling the complex parsing is relatively straightforward once some utility classes are written. However, handling the statefulness is harder, especially given that an SFU always needs to be prepared for packet loss and reordering, so packets that use some state may arrive before the packet that provides it. Thus this work needs to keep track of the state that’s reported in those packets that carry it, and pass it forward to the parser to parse subsequent headers, while handling the possibility that a state update may have been missed.
Because the AV1 DD is a header extension, it can be applied to codecs other than AV1 itself. Notably, this allows us to indicate temporal scalability of an H.264 stream, which is useful because H.264 (without its little-implemented SVC extensions) has no way to indicate temporal scalability of its packets. As a result, the work to support the AV1 DD also allows Jitsi to enable scalable H.264 streams for JVB-mediated conferences as well! (Though there are currently some bugs in Chrome that make this less efficient than it could be – see here and here.) In fact the header can be applied to any video codec, though we still prefer to handle VP8 and VP9 using their in-payload information when that information is available.
Another notable feature of the AV1 dependency descriptor is its concept of “decode targets”, which are a specific layer that a decoder can decode, and thus that an SFU can forward. These usually correspond to a specific spatial and temporal layer, but technically they do not have to. The idea is that a decoder can choose among the various decode targets present in the stream. In most cases it would want to choose the highest-quality one available, but in some circumstances (for instance if it was CPU-constrained, or displaying a source in a small window) it could choose a lower-quality target instead.
This has the consequence that a stream needs to be explicit about what decode targets are actually present. An SFU, by design, forwards only some of the layers in a stream; this is what it means to “selectively forward”. As a result, the ultimate decoder needs to know what layers it can actually expect to receive, vs. which ones it won’t receive, or it can end up waiting forever while it never gets the frames it thinks it should render. To handle this case, the AV1 dependency descriptor contains a decode target bitmask in the AV1 header extension to indicate which layers are still present in the stream. This bitmask then needs to be updated every time the SFU changes its layer forwarding decision, so the decoder doesn’t try to wait to decode a decode target that won’t be arriving any more, or, conversely, so it can know to start decoding a new layer that’s newly started arriving. Fortunately, the logic to do this work is not too complicated, and is similar in complexity to the logic needed to modify the in-payload information for forward VP8 or VP9 streams.
With the release of Chromium M111, significant advancements were made in WebRTC, particularly with the introduction of Scalable Video Coding (SVC) support. This update enabled WebRTC applications to configure encoding parameters for SVC by extending the RTCRtpEncodingParameters dictionary. Around the same time, Chrome also introduced AV1 and VP9 simulcast support. Before Chromium M111, the K-SVC mode (Keyframe-scalable video coding) was the only supported mode for SVC. This update allowed Jitsi Meet to experiment with various scalability modes for AV1 and VP9.
In a Jitsi conference, the client and JVB work in tandem to ensure efficient video streaming. This involves an ongoing exchange of sender and receiver video constraints.
This dynamic coordination minimizes bandwidth usage and optimizes network resources while maintaining the quality of the user experience. Once the client receives sender constraints, it configures its outbound video streams using RTCRtpEncodingParameters. These parameters are tailored based on:
For AV1 and VP9, three operating modes were tested:
SVC allows a single video stream to be encoded in layers, each layer adding more detail or resolution to the base stream while simulcast involves sending multiple independent video streams of the same content, each at a different resolution and bitrate.
| Aspect | Simulcast | SVC |
Encoding |
Multiple streams encoded separately. |
One stream encoded with multiple layers. |
Bandwidth Usage (Sender) |
Higher (multiple streams). |
Lower (single stream). |
CPU Usage (Sender) |
High (due to multiple encodings). |
Lower (single encoding with layers). |
CPU Usage (Receiver) |
Lower (no need to decode layers). |
Higher (decoding layered streams). |
| Adaptability | Coarser (switching between streams). |
Finer (dynamic layer adjustment). |
Compatibility |
Broadly supported in WebRTC platforms. |
Limited support, requires advanced codecs. |
After extensive performance testing and careful evaluation of product requirements, Jitsi selected full SVC mode as the default configuration for both AV1 and VP9. This choice ensures optimal scalability and video quality across Jitsi’s deployments. However, this behavior is not rigid; it is configurable and can be easily overridden through config.js settings, providing flexibility to adapt to specific use cases or deployment needs.
To determine the optimal video codec for use in the Jitsi Meet client, Jitsi conducted comprehensive testing under realistic conditions to ensure that codec selection would meet product needs for quality, performance, and scalability. Below is an overview of the methodologies and considerations involved:
By rigorously analyzing these metrics, Jitsi optimized its codec selection strategy. While Full SVC modes remained the default for high-performance scenarios, fallback options like VP9 and VP8 were configured for legacy or resource-constrained devices. This comprehensive approach ensures that the Jitsi Meet client provides the best possible video experience across a wide range of devices and network conditions.
With the integration of AV1 into Jitsi Meet, users benefit from superior compression and high-quality video at lower bitrates. However, these advantages come at the cost of increased computational demands, especially on low-end devices. To address this, Jitsi introduced a three-fold adaptive quality control mechanism, ensuring a seamless experience even under CPU constraints.
This adaptive approach enables Jitsi Meet to leverage the advanced capabilities of AV1 while ensuring that users with diverse hardware configurations can participate in meetings without disruptions caused by excessive CPU usage.
However, when the CPU spike originates from an external process rather than the Jitsi Meet client, the adaptive mode ensures that quality degradation is minimal. To enhance the user experience, Jitsi Meet also incorporates a recovery mechanism that restores the video configuration once the external constraints are resolved.
This gradual approach minimizes the risk of overloading the system during recovery. It also adapts to fluctuating CPU availability, maintaining a balance between performance and quality. The client handles this entire process dynamically without any user interaction, providing a seamless experience.
Firefox and Safari do not advertise support for the AV1 codec yet. As a result, when users on these browsers join a call, all other participants automatically switch to the next codec in the preferred list, ensuring compatibility across all endpoints.
Additionally, while Chromium-based mobile endpoints are capable of both encoding and decoding AV1, Jitsi has opted to use AV1 only for decoding. For encoding, a lower-complexity codec is used, as encoding typically imposes a higher CPU load compared to decoding. This decision balances performance and device resource constraints, especially on mobile devices.
We have great news for you then!! AV1 support was introduced to Jitsi in June 2024 and has been available in our stable packages ever since. Initially, AV1 had to be manually configured as the preferred codec through config.js settings, allowing users to opt in.
Building on this, AV1 was soon made the preferred codec on meet.jit.si, marking a significant step in leveraging its advanced compression capabilities. Starting with release stable-9909, AV1 became the default preferred codec in our Docker deployments, ensuring out-of-the-box support for users opting for containerized setups.
After thorough experimentation and analysis of real-world performance data, we’re excited to share that AV1 will very soon become the default preferred codec in all deployments, bringing its exceptional bandwidth efficiency and video quality to a broader audience. Stay tuned!
Your personal meetings team.
P.S – With contributions from Jonathan Lennox (Jitsi VideoBridge)
The post AV1 and more … how does Jitsi Meet pick video codecs? appeared first on Jitsi.
Sangoma TeamHub—a user-friendly app designed to simplify team collaboration and boost productivity—is now available to new users of Sangoma's UCaaS platform CommUnity.
The post Introducing Sangoma TeamHub for CommUnity appeared first on Sangoma Technologies.
A portion of the telephone network that dates back several decades is often called Plain Old Telephone Service, or POTS for short. In recent years.
The post How to Successfully Migrate from Traditional Telecom to Modern Packet-Based Phone Networks appeared first on Sangoma Technologies.
Tuesday, December 10 @ 5PM CET / 11AM ET / 8AM PT / 16:00 UTC Join Chad Hart, Editor of webrtcHacks, for an analysis of WebRTC trends in GitHub, StackOverflow, and other open-source communities. Leveraging advanced quantitative analysis techniques, this talk examines millions of GitHub events and developer activity data to uncover key trends […]
The post Upcoming Livestream 10-Dec: 2024 WebRTC in Open Source Review appeared first on webrtcHacks.
Sangoma TeamHub—a user-friendly app designed to simplify team collaboration and boost productivity—is now available to new users of Sangoma’s UCaaS platform CommUnity.
With features like voice calling, real-time chat, SMS, video meetings, and file storage, you get all your communication tools in one place without jumping between multiple apps. Stay connected wherever you are! Access Sangoma TeamHub easily through your web browser or use the desktop app on Mac or Windows. It’s also available on iOS and Android mobile devices.
Key Features:
Talk: Make clear calls using Sangoma TeamHub’s built-in softphone, which features call transfers and rings across all your devices. Enjoy convenient access to visual voicemail and the ability to seamlessly hand off calls, allowing you to continue conversations on another device. Fine-tune your audio settings to fully leverage CommUnity’s powerful voice features, including parking lots, advanced ‘Find Me/Follow Me’ routes, and your own conference bridge.
Sangoma TeamHub focuses on user satisfaction by providing robust and secure communications, as well as easy customization. Accessible from any internet-connected device, it offers timely notifications whether you’re in the office, working remotely, or on the go, keeping you connected and informed.
Sangoma TeamHub is the ultimate tool for team collaboration and communication, helping you work efficiently from anywhere.
Learn more about mastering teamwork with Sangoma TeamHub today!
Join us on December 12th for the exclusive TeamHub for CommUnity webinar: Register Now!
The post Introducing Sangoma TeamHub for CommUnity appeared first on Sangoma Technologies.
As businesses increasingly migrate to IP networks, a significant challenge remains: how do you transition from legacy POTS (Plain Old Telephone Service) while maintaining the investment in devices like phones, fax machines, and security equipment?
The post Seamlessly Transition from POTS to IP Networks with Sangoma Vega Gateways appeared first on Sangoma Technologies.
In a significant shift in the Unified Communications (UC) landscape, NEC announced its decision to exit the UCaaS market, in addition to its earlier announcement to leave the on-prem market. These moves have left many NEC partners and prospects wondering, “Where do we go from here?” NEC in the News Intermedia Cloud Communications and NEC […]
The post NEC Steps Out of the UCaaS Market, Now What? appeared first on Sangoma Technologies.
As part of our commitment to innovation and customer support, we are excited to announce the launch of @ASKSangoma, an AI-powered Knowledge Bot. The latest version 3.3 of the Sangoma Teamhub collaboration app introduces a new bot designed to assist our partners and customers with inquiries regarding Sangoma’s product portfolio. Trained on comprehensive product documentation, […]
The post Introducing ASK Sangoma: Your AI-powered Knowledge Bot for Quicker Support appeared first on Sangoma Technologies.
WebRTC’s peer connection includes a getStats method that provides a variety of low-level statistics. Basic apps don’t really need to worry about these stats but many more advanced WebRTC apps use getStats for passive monitoring and even to make active changes. Extracting meaning from the getStats data is not all that straightforward. Luckily return author […]
The post Power-up getStats for Client Monitoring appeared first on webrtcHacks.
ClueCon has just finished. Here are some of my photos.
Seven Du from China won the Macbook. Congratulations.
ClueCon has just finished. Here are some of my photos.
Seven Du from China won the Macbook. Congratulations.
After upgrading to Ubuntu Jammy and Asterisk 18.10, I saw the following messages in my logs:
WARNING[360166]: loader.c:2487 in load_modules: Module 'chan_sip' has been loaded but was deprecated in Asterisk version 17 and will be removed in Asterisk version 21.
WARNING[360174]: chan_sip.c:35468 in deprecation_notice: chan_sip has no official maintainer and is deprecated. Migration to
WARNING[360174]: chan_sip.c:35469 in deprecation_notice: chan_pjsip is recommended. See guides at the Asterisk Wiki:
WARNING[360174]: chan_sip.c:35470 in deprecation_notice: https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
WARNING[360174]: chan_sip.c:35471 in deprecation_notice: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip
and so I decided it was time to stop postponing the
overdue migration
of my working setup from
chan_sip
to
res_pjsip.
It turns out that it was not as painful as I expected, though the conversion script bundled with Asterisk didn't work for me out of the box.
Before you start, one very important thing to note is that the SIP debug
information you used to see when running this in the asterisk console
(asterisk -r):
sip set debug on
now lives behind this command:
pjsip set logger on
The first thing I migrated was the config for my two SIP phones (Snom 300 and Snom D715).
The original config for them in sip.conf was:
[2000]
; Snom 300
type=friend
qualify=yes
secret=password123
encryption=no
context=full
host=dynamic
nat=no
directmedia=no
mailbox=10@internal
vmexten=707
dtmfmode=rfc2833
call-limit=2
disallow=all
allow=g722
allow=ulaw
[2001]
; Snom D715
type=friend
qualify=yes
secret=password456
encryption=no
context=full
host=dynamic
nat=no
directmedia=yes
mailbox=10@internal
vmexten=707
dtmfmode=rfc2833
call-limit=2
disallow=all
allow=g722
allow=ulaw
and that became the following in pjsip.conf:
[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0
external_media_address = myasterisk.dyn.example.com
external_signaling_address = myasterisk.dyn.example.com
local_net = 192.168.0.0/255.255.0.0
[2000]
type = aor
max_contacts = 1
[2000]
type = auth
username = 2000
password = password123
[2000]
type = endpoint
context = full
dtmf_mode = rfc4733
disallow = all
allow = g722
allow = ulaw
direct_media = no
mailboxes = 10@internal
auth = 2000
outbound_auth = 2000
aors = 2000
[2001]
type = aor
max_contacts = 1
[2001]
type = auth
username = 2001
password = password456
[2001]
type = endpoint
context = full
dtmf_mode = rfc4733
disallow = all
allow = g722
allow = ulaw
direct_media = yes
mailboxes = 10@internal
auth = 2001
outbound_auth = 2001
aors = 2001
The different direct_media line between the two phones has to do with how
they each connect to my Asterisk
server
and whether or not they have access to the Internet.
For some reason, my internal calls (from one SIP phone to the other) didn't
work when using "aliases". I fixed it by changing this blurb in
extensions.conf from:
[speeddial]
exten => 1000,1,Dial(SIP/2000,20)
exten => 1001,1,Dial(SIP/2001,20)
to:
[speeddial]
exten => 1000,1,Dial(${PJSIP_DIAL_CONTACTS(2000)},20)
exten => 1001,1,Dial(${PJSIP_DIAL_CONTACTS(2001)},20)
I have not yet dug into what this changes or why it's necessary and so feel free to leave a comment if you know more here.
Once I had the internal phones working, I moved to making and receiving phone calls over the PSTN, for which I use VoIP.ms with encryption.
I had to change the following in my sip.conf:
[general]
register => tls://555123_myasterisk:password789@vancouver2.voip.ms
externhost=myasterisk.dyn.example.com
localnet=192.168.0.0/255.255.0.0
tcpenable=yes
tlsenable=yes
tlscertfile=/etc/asterisk/asterisk.cert
tlsprivatekey=/etc/asterisk/asterisk.key
tlscapath=/etc/ssl/certs/
[voipms]
type=peer
host=vancouver2.voip.ms
secret=password789
defaultuser=555123_myasterisk
context=from-voipms
disallow=all
allow=ulaw
allow=g729
insecure=port,invite
canreinvite=no
trustrpid=yes
sendrpid=yes
transport=tls
encryption=yes
to the following in pjsip.conf:
[transport-tls]
type = transport
protocol = tls
bind = 0.0.0.0
external_media_address = myasterisk.dyn.example.com
external_signaling_address = myasterisk.dyn.example.com
local_net = 192.168.0.0/255.255.0.0
cert_file = /etc/asterisk/asterisk.cert
priv_key_file = /etc/asterisk/asterisk.key
ca_list_path = /etc/ssl/certs/
method = tlsv1_2
[voipms]
type = registration
transport = transport-tls
outbound_auth = voipms
client_uri = sip:555123_myasterisk@vancouver2.voip.ms
server_uri = sip:vancouver2.voip.ms
[voipms]
type = auth
password = password789
username = 555123_myasterisk
[voipms]
type = aor
contact = sip:555123_myasterisk@vancouver2.voip.ms
[voipms]
type = identify
endpoint = voipms
match = vancouver2.voip.ms
[voipms]
type = endpoint
context = from-voipms
disallow = all
allow = ulaw
allow = g729
from_user = 555123_myasterisk
trust_id_inbound = yes
media_encryption = sdes
auth = voipms
outbound_auth = voipms
aors = voipms
rtp_symmetric = yes
rewrite_contact = yes
send_rpid = yes
timers = no
dtmf_mode = rfc4733
The TLS method line is needed since the default in Debian OpenSSL is too
strict. The timers
line is to prevent outbound calls from getting dropped after 15 minutes.
Finally, I changed the Dial() lines in these extensions.conf blurbs from:
[from-voipms]
exten => 5551231000,1,Goto(2000,1)
exten => 2000,1,Dial(SIP/2000&SIP/2001,20)
exten => 2000,n,Goto(in2000-${DIALSTATUS},1)
exten => 2000,n,Hangup
exten => in2000-BUSY,1,VoiceMail(10@internal,su)
exten => in2000-BUSY,n,Hangup
exten => in2000-CONGESTION,1,VoiceMail(10@internal,su)
exten => in2000-CONGESTION,n,Hangup
exten => in2000-CHANUNAVAIL,1,VoiceMail(10@internal,su)
exten => in2000-CHANUNAVAIL,n,Hangup
exten => in2000-NOANSWER,1,VoiceMail(10@internal,su)
exten => in2000-NOANSWER,n,Hangup
exten => _in2000-.,1,Hangup(16)
[pstn-voipms]
exten => _1NXXNXXXXXX,1,Set(CALLERID(all)=Francois Marier <5551231000>)
exten => _1NXXNXXXXXX,n,Dial(SIP/voipms/${EXTEN})
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Set(CALLERID(all)=Francois Marier <5551231000>)
exten => _NXXNXXXXXX,n,Dial(SIP/voipms/1${EXTEN})
exten => _NXXNXXXXXX,n,Hangup()
exten => _011X.,1,Set(CALLERID(all)=Francois Marier <5551231000>)
exten => _011X.,n,Authenticate(1234)
exten => _011X.,n,Dial(SIP/voipms/${EXTEN})
exten => _011X.,n,Hangup()
exten => _00X.,1,Set(CALLERID(all)=Francois Marier <5551231000>)
exten => _00X.,n,Authenticate(1234)
exten => _00X.,n,Dial(SIP/voipms/${EXTEN})
exten => _00X.,n,Hangup()
to:
[from-voipms]
exten => 5551231000,1,Goto(2000,1)
exten => 2000,1,Dial(PJSIP/2000&PJSIP/2001,20)
exten => 2000,n,Goto(in2000-${DIALSTATUS},1)
exten => 2000,n,Hangup
exten => in2000-BUSY,1,VoiceMail(10@internal,su)
exten => in2000-BUSY,n,Hangup
exten => in2000-CONGESTION,1,VoiceMail(10@internal,su)
exten => in2000-CONGESTION,n,Hangup
exten => in2000-CHANUNAVAIL,1,VoiceMail(10@internal,su)
exten => in2000-CHANUNAVAIL,n,Hangup
exten => in2000-NOANSWER,1,VoiceMail(10@internal,su)
exten => in2000-NOANSWER,n,Hangup
exten => _in2000-.,1,Hangup(16)
[pstn-voipms]
exten => _1NXXNXXXXXX,1,Set(CALLERID(all)=Francois Marier <5551231000>)
exten => _1NXXNXXXXXX,n,Dial(PJSIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Set(CALLERID(all)=Francois Marier <5551231000>)
exten => _NXXNXXXXXX,n,Dial(PJSIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011X.,1,Set(CALLERID(all)=Francois Marier <5551231000>)
exten => _011X.,n,Authenticate(1234)
exten => _011X.,n,Dial(PJSIP/${EXTEN}@voipms)
exten => _011X.,n,Hangup()
exten => _00X.,1,Set(CALLERID(all)=Francois Marier <5551231000>)
exten => _00X.,n,Authenticate(1234)
exten => _00X.,n,Dial(PJSIP/${EXTEN}@voipms)
exten => _00X.,n,Hangup()
Note that it's not just replacing SIP/ with PJSIP/, but it was also
necessary to use a format supported by
pjsip
for the channel since SIP/trunkname/extension isn't supported by pjsip.
GStreamer is one of the oldest and most established libraries for handling media. As a core media handling element in Linux and WebKit that as launched near the turn of the century, it is not surprising that many early WebRTC projects use various pieces of it. Today, GStreamer has expanded options for helping developers plumb […]
The post WebRTC Plumbing with GStreamer appeared first on webrtcHacks.
In the evolving landscape of virtual meetings, seamless connectivity remains paramount. SIP integration enables participants to join meetings from various devices, including hardware phones or softphones such as Bria, video conferencing systems such as Zoom, and traditional telephony systems. This broadens the scope of participants who can connect to Jitsi conferences, making it more inclusive.
Note: We are using JaaS here for the purpose of simplicity, but all of this can be deployed using the Open Source components available on GitHub.
Note: Replace pinCode with the specific conference PIN provided for your Jitsi conference
SIP audio-only connectivity provides a cost-effective and reliable way for participants to join Jitsi conferences. It reduces bandwidth consumption and costs, making it ideal in scenarios where video isn’t really needed, such as webinars. This option ensures users with limited internet access or slower connections can participate without interruptions.
Integrating Voximplant with Jitsi Meet involves several key steps:
Note: Voximplant can be replaced with a programmable SIP server such as Kamailio or OpenSIPS.
While SIP is these days referred to as legacy, it remains the most used protocol for VoIP and acts as the common denominator across many vendors in the industry. Thus it’s a great candiate for connecting anything to everything 
Your personal meetings team.
The post Connecting anything to everything via SIP appeared first on Jitsi.
The Risks of Sticking with Unsupported On-Premises PBX Systems and Why Switching to Sangoma’s Switchvox is Essential Now You’ve found yourself in a tough spot – change is not just coming; it’s here. NEC has announced its departure from the on-premises PBX market, shifting its focus towards cloud-based solutions. So where does that leave you? […]
The post Delivering a One-two Punch to NEC’s Departure from the On-premises PBX Market appeared first on Sangoma Technologies.
The Risks of Sticking with Unsupported On-Premises PBX Systems and Why Switching to Sangoma’s Switchvox is Essential Now You’ve found yourself in a tough spot – change is not just coming; it’s here. NEC has announced its departure from the on-premises PBX market, shifting its focus towards cloud-based solutions. So where does that leave you? […]
The post Delivering a One-two Punch to NEC’s Departure from the On-premises PBX Market appeared first on Sangoma Technologies.
Maximizing stream quality on an imperfect network in real-time is a delicate balancing act. If you send too much information then will cause congestion and packet loss. If you send too little then your video quality (or audio) will look like garbage. But how much can you send? One of the techniques used to find […]
The post Probing WebRTC Bandwidth Probing – why and how in gcc appeared first on webrtcHacks.
As of 2023, a significant portion of businesses still rely on on-premises systems. While the specific percentage of businesses using on-premises UC solutions is not detailed in recent reports, it’s evident that both on-premises and cloud solutions continue to coexist, with each offering distinct advantages depending on organizational needs and priorities. The unified communications market […]
The post Why Choose On-Premises Solutions In a Cloud-Dominated Era? appeared first on Sangoma Technologies.
As of 2023, a significant portion of businesses still rely on on-premises systems. While the specific percentage of businesses using on-premises UC solutions is not detailed in recent reports, it’s evident that both on-premises and cloud solutions continue to coexist, with each offering distinct advantages depending on organizational needs and priorities. The unified communications market […]
The post Why Choose On-Premises Solutions In a Cloud-Dominated Era? appeared first on Sangoma Technologies.
In the last stable release, Jitsi enabled a new feature called SSRC rewriting that improves the system performance for very large calls. This feature helps reduce the overall load on the system by reducing the number of signaling messages that get exchanged during a large call involving hundreds of endpoints. It also reduces the load on the local endpoint drastically by restricting the number of audio and video decoders created by the WebRTC engine thereby offering a better user experience for large calls.
When this feature is enabled, only a fixed set (let’s say up to 50) of SSRCs are signaled to the downstream endpoints in the call irrespective of the call size. An SSRC is nothing but a unique ID used for identifying a stream of RTP packets that belong to an audio or a video source. When additional media sources are requested by the receiver, the Jitsi Videobridge (JVB) overwrites the SSRCs of the newly requested media streams with that of the ones that were already signaled to the client before and are no longer needed. Therefore, no more than 50 SSRCs need to be signaled to the endpoints even if the number of media sources that will be routed in total far exceeds the set limit.
Moving on to the why – what is the problem that we were trying to solve by implementing SSRC rewriting?
The challenges that we faced when adding support for very large calls with respect to source signaling with the existing approach of signaling every new source to every other participant in the call were two fold.
At the client level, the number of m-lines in the SDP grew linearly with every remote source that got added to the call irrespective of whether media for that particular source gets routed to the endpoint or not. When a new m-line with SSRC is added to the remote SDP, the libwebrtc engine creates a transceiver and does all the plumbing necessary to decode a media stream with the given SSRC if and when it starts receiving it from the JVB. This tied up resources on the local endpoint unnecessarily and introduced delays in renegotiation cycles which resulted in unpleasant user experience. The client can also hit transceiver and SDP parser limits imposed by the browser resulting in unexpected behaviors. These performance issues are more pronounced on mobile endpoints which have fewer resources to begin with compared to the endpoints running on desktops.
On the backend, as the number of participants grew, so did the number of audio and video sources that needed to be signaled to every other participant in the call. This made the signaling traffic from prosody (XMPP communication server) to the endpoints grow quadratically. This was a problem, because Prosody, which is single-threaded, was already the bottleneck when scaling calls. Previously we had to introduce artificial delays in signaling in order to reduce the load. This caused long delays for the media to be established across participants when they unmuted their audio or video for the first time and was very disruptive to large meetings.
The solution to both of these problems was to switch to a demand based signaling mechanism where only a limited number of remote audio and video tracks are signaled to the endpoints depending on what is being needed or requested by them in real time instead of signaling all the known media sources in the call as and when they get added to the call.
Jitsi Videobridge (JVB) uses a slightly different approach for audio and for video when determining what sources to forward. Forwarding decisions for audio are based simply on the “loudness” of the streams determined from the audio level RTP extension.
With SSRC rewriting, JVB uses a separate SSRC space for each receiver. It maintains a map from an SSRC number to the name of a source. Changes to the map are signaled to the receiver over the direct signaling channel (a WebRTC DataChannel over SCTP), using partial updates:
[modules/RTC/BridgeChannel.js] <e.onmessage>: Received AudioSourcesMap: [{"source":"fc63db0f-a0","owner":"fc63db0f","ssrc":2602882473},{"source":"449360a0-a0","owner":"449360a0","ssrc":1358697798}]
[modules/RTC/BridgeChannel.js] <e.onmessage>: Received VideoSourcesMap: [{"source":"e01f2103-v0","owner":"e01f2103","ssrc":3129389873,"rtx":3219602897,"videoType":"CAMERA"},{"source":"9ac8fef2-v0","owner":"9ac8fef2","ssrc":1542056973,"rtx":1571329554,"videoType":"CAMERA"},{"source":"ed6b60f5-v0","owner":"ed6b60f5","ssrc":550523896,"rtx":2808127984,"videoType":"CAMERA"}]When a new stream needs to be forwarded, it is allocated an SSRC. Before the limit is reached, JVB simply generates a new SSRC number, and when the limit has been reached the oldest entry is reused. Let’s look at an example to make this more clear. Assume the limit is set to just 3, and the available sources are A, B, C, D, E. Initially the map is empty. When A starts sending packets, we allocate SSRC 101 to it and signal it to the receiver like this:
AudioSourcesMap: [{"source":"A","owner":"endpoint-A","ssrc":101}]Similarly when B and C start to speak we allocate SSRCs 102 and 103 for them:
AudioSourcesMap: [{"source":"B","owner":"endpoint-B","ssrc":102}]AudioSourcesMap: [{"source":"C","owner":"endpoint-C","ssrc":103}]Now we have reached the limit of 3 SSRCs. When D starts to speak, we’ll find the source in the map that has been active least recently (let’s say that’s B) and re-use its SSRC for D. We’ll signal an update (“SSRC 102 now belongs to D”):
AudioSourcesMap: [{"source":"D","owner":"endpoint-D","ssrc":102}]The scheme is identical for video, except the forwarding decisions are made in a different way. Receivers explicitly signal their preferences using video constraints. The source names and their mute status are published in presence when an endpoint signals its source information to the Jitsi Conference Focus (Jicofo) and therefore this information is already available with all the other endpoints in the call. Based on the current layout in the UI and the user’s preferences, the client sends the updated receiver video constraints over the bridge channel.
A bandwidth allocation algorithm in the JVB then decides which streams to forward to a particular receiver, based on its constraints and current network conditions:
[modules/RTC/BridgeChannel.js] <Fa.sendReceiverVideoConstraintsMessage>: Sending ReceiverVideoConstraints with {"constraints":{"ed6b60f5-v0":{"maxHeight":360},"e01f2103-v0":{"maxHeight":360},"9ac8fef2-v0":{"maxHeight":360}},"defaultConstraints":{"maxHeight":0},"lastN":-1,"onStageSources":[],"selectedSources":[]}On receiving an update to one of the maps (audio or video), the client adds the signaled SSRCs to the remote description on the peerconnection. The browser then fires a track event for each of the SSRCs, the corresponding remote tracks are then added to the HTMLElements associated with the remote user.
So when the audio packets with this SSRC arrive, the browser starts decoding the media and plays it through the selected audio output device. If the SSRC is already in use (i.e. the limit on the bridge has been reached) the client updates the owner of the associated track so that it gets attached to the corresponding HTMLAudioElement and the audio switches over to the new speaker seamlessly.
The video track creation process is the same as that of the audio tracks as described above. The client application needs to update the track’s owner whenever there is an updated source map involving the SSRC that is assigned to the track and re-attach it to the corresponding HTMLVideoElement so that the correct video stream is rendered in the remote participant’s viewport.
But wait, re-using an SSRC like this is okay for audio because the streams simply get mixed before playback, but what about video, how do we avoid video content being rendered in the wrong viewport when signaling and media race? That’s the elegance of this approach, we simply use a large enough limit (larger than the maximum number of streams forwarded at any one time) and the occurrence becomes extremely unlikely. If the limit is larger by K, then K new forwarding decisions must be made before the signaling arrives at the receiver for the problem to happen.
So how do we choose the limits? We have the constraint just mentioned, but also an interesting trade-off. If the limit is too high we’re using unnecessary resources at the receivers. But if the limit is too low, we’ll be signaling updates more often. We have chosen to set the limits to 50 by default. That’s 50 for audio and 50 for video, which is well above the maximum of 25 tiles that we display at any time.
When SSRC rewriting is enabled, the number of source signaling messages can increase drastically based on the SSRC limits set for the conference and the total number of participants in the call. Imagine a call with 100 participants where everyone has their video on; UI shows a grid of 25 participants and the SSRC limit is set to 25. Whenever the user scrolls to the next grid of 25 participants, existing SSRCs get remapped. This happens everytime the user scrolls back and forth. This results in a lot of signaling messages over the bridge channel. What if the websocket connection for the bridge channel is down at this time? This would result in videos not being rendered or audio from new dominant speakers not being heard which can be very disruptive to meetings. All the sources are signaled immediately after the websocket connection reforms but even minimal disruptions to audio can be very annoying.
To mitigate these issues, Jitsi client switches to using WebRTC’s SCTP data channel for establishing the bridge channel instead of using a websocket. This ensures that the bridge channel is up and running all the time as long as the media connection between the client and the JVB is up. This results in minimal or no disruptions to the signaling messages from the JVB to the downstream endpoints.
This feature has been well tested and has been running on meet.jit.si for the past few months now, with limits set to 50. We also enabled it by default in our last stable release of the Debian packages and Docker images. We will be releasing it soon to all our production deployments in the next few releases pending investigation into some SCTP crashes that we are seeing in the JVB .
Your personal meetings team.
The post Improving performance on very large calls: introducing SSRC rewriting appeared first on Jitsi.
In a digital age where the Unified Communications (UC) landscape is rapidly expanding, On-Premises UC solutions stand out as the bedrock for businesses aiming to secure, customize, and control their communication infrastructures. With the UC market projected to swell to USD 145.58 billion by 2024 and further grow to an impressive USD 496.30 billion by […]
The post Navigating the Future with Confidence: The Strategic Imperative of On-Premises UC Solutions appeared first on Sangoma Technologies.
In a digital age where the Unified Communications (UC) landscape is rapidly expanding, On-Premises UC solutions stand out as the bedrock for businesses aiming to secure, customize, and control their communication infrastructures. With the UC market projected to swell to USD 145.58 billion by 2024 and further grow to an impressive USD 496.30 billion by […]
The post Navigating the Future with Confidence: The Strategic Imperative of On-Premises UC Solutions appeared first on Sangoma Technologies.
The rumor mill started churning when cloud-based solutions began gaining prominence. These solutions offer scalable, flexible alternatives to traditional on-premises setups. As more companies migrated to the cloud, a narrative emerged suggesting that on-premises solutions like Switchvox were becoming obsolete. Though popular, this view only captures part of the picture. The Reality of Switchvox On-Premises […]
The post Switchvox On-Premises: Alive & Well appeared first on Sangoma Technologies.
The rumor mill started churning when cloud-based solutions began gaining prominence. These solutions offer scalable, flexible alternatives to traditional on-premises setups. As more companies migrated to the cloud, a narrative emerged suggesting that on-premises solutions like Switchvox were becoming obsolete. Though popular, this view only captures part of the picture. The Reality of Switchvox On-Premises […]
The post Switchvox On-Premises: Alive & Well appeared first on Sangoma Technologies.