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In traditional broadcast SDI facilities, we use a point-to-point distribution system with one-to-one mapping. That is, to connect the output of a camera to a production switcher we take the SDI output of the camera and connect it directly to one of the production switchers inputs. If we then want to monitor the camera feed going to the production switcher, we must either disconnect the input or introduce a Distribution Amplifier (DA) which has multiple outputs. Each one of these then forms a one-to-one mapping with monitors, routing matrices and other processing equipment.

By introducing a DA, we have effectively provided a one-to-many mapping system.

In the IP world, to provide one-to-many mappings, we substitute the physical DA with an abstracted logical model using IP routers. Instead of having individual SDI cables connecting the equipment together, we take advantage of the IP-router’s ability to be able to duplicate packets and create a multicast system.

IP routing generally works on the unicast model; when a computer requests data from a server, there will be just one source and one destination. However, if a playout server wants to send an online movie to many users, a mapping of one-to-many is used and this is called multicast.

Multicasting

Multicasting is much more flexible than SDI infrastructures using matrices and DA’s as we are not limited to working in the SDI domain. If a user wants to watch the studio output from their office, or the green room, then we don’t have to worry about running new SDI cables to them, or adding the studio to RF distribution; we just set up another multicast user in the IP-router.

The IP-router provides the one-to-many mapping by distributing the video at the IP packet level. Each IP datagram is copied and then sent out on the appropriate port of the router to the user.

The source device, such as camera, production switcher or sound console, has no knowledge of which devices are receiving their output in an analogous way to traditional broadcast equipment. The sound console has no knowledge of the destination of its outputs, only that they are seeing a high impedance load to a Distribution Address at the end of a twisted pair. The Ethernet output of a camera has no knowledge of where its packets are going and can only see the network interface card of the IP-router or switcher it is connected to.

An alternative method to multicasting would be to set the camera’s destination address of each device requiring the stream. This would cause two problems. Firstly, the overhead of having to set the IP destination addresses of six cameras in a studio, each connected to potentially ten different destinations, is an administrative nightmare and would be unworkable. Secondly, we would increase the network load by the number of destinations set in the camera’s routing table, as each packet would be individually sent to each device requiring the stream.

Multicasting solves both problems as it relies on the receiving equipment actively opting in to the feed, and IP packets are only sent once to each group destination, with the routers providing duplication only on the network branches that need the stream. There are three key concepts in a multicasting system: the group address, reverse path forwarding and IGMP.

IGMP Opt-In

The multicast group address is the destination IP address of the equipment creating the audio or video stream, i.e., a camera or microphone. Each camera, microphone, production switcher output or sound console aux send will require its own group address. The multicast group addresses are special IP addresses constrained in the range 224.0.0.0 to 239.255.255.255, approximately 248 million groups. Some of these are reserved and a full list can be found at the Internet Assigned Numbers Authority (IANA).

Internet Group Membership Protocol (IGMP) provides the mechanism for equipment receiving the stream (monitors etc.) to tell its router that it wants to receive a particular group. Each IP-router passes this information along the line to the source equipment so that only the networks that require the stream relay it, thus reducing network congestion.

In figure 2 the router will periodically send an IGMP Host Membership Query message to all devices connected to its networks, in this case the sound console responds with an IGMP Host Report Group 1 and Group 2 message, advising the router that it wants to receive Mic-1 and Mic-2 audio streams. Mic-3 (Group 3) is not sent to the sound console as the sound console did not opt-in to receive that group.

Unicast routing works by looking at the destination IP address of the packet and comparing it to its look-up table so the router knows where to forward the packet to. Multicast routing uses the source address in the IP-datagram to route back to find out where the source of the packet is located, a system called Reverse Path Forwarding (RPF).

The beauty of IP networks in general, and multicasting specifically, is that the data can move in both directions simultaneously. For example, a sound console could have all its aux sends and returns on one Ethernet cable, potentially saving tons of twisted pair cable. The sound console becomes both a multicast receiver and multicast broadcaster. Assuming the availability of sufficient link capacity, IP routers can be used with the production switcher to achieve the same multicasting with its video inputs and outputs, with hundreds of feet of coax cable being substituted by CAT6 Ethernet and fiber optic cable.

If a producer wants to watch the studio output in their office, their desktop computer or tablet can be easily configured to receive the group address of the studio output. No configuration will be required by the IT department, and no additional cables will need to be run to their office. However, this introduces the topic of network security. It might be that the producers do not want the whole station to be able to see the studio output, so the routers would have to be configured to stop them sending certain streams to certain networks.

All The SMPTE Standards

In broadcast studios, the adoption of SMPTE’s ST 2110 suite of standards has become the defacto standard for both greenfield developments and those transitioning to IP from baseband infrastructures. By using the User Datagram Protocol (UDP), video and audio streams are divided into IP packets, creating a high-throughput, fire-and-forget transport system. This transition to a fully IP-based infrastructure brings greater scalability, flexibility, and efficiency to production environments, while dramatically reducing costs compared to traditional SDI-based systems. As point-to-point cabling is replaced by standard enterprise network technology, production facilities can distribute uncompressed video, audio, and data independently across the network, unlocking a new level of operational agility.

ST 2110 extends well beyond video and audio. The standard also includes provisions for metadata and control streams, allowing for the synchronized transport of all essential production data. While metadata and control flows often employ acknowledgement-based protocols like the Transmission Control Protocol (TCP) for reliability, the video and audio elements typically rely on Forward Error Correction (FEC) to detect and recover from burst-type errors. This ensures consistent data integrity while preserving the high throughput and low latency that broadcast operations demand.

One of the major strengths of ST 2110 is its ability to distribute uncompressed media with minimal latency. Video, audio, and ancillary data are carried as separate, precision timed streams, each maintaining full broadcast quality. This separation gives engineers the freedom to manage and process each element independently, for instance, adjusting or rerouting audio without touching the video, or handling metadata without disrupting live feeds.

Scalability and flexibility are greatly enhanced. Adding new equipment, such as a camera or microphone, often requires nothing more than connecting the device to the network rather than installing additional SDI or AES cabling and expanding router hardware. However, this increased flexibility comes with the need for careful network design. Engineers must consider link capacity, IP address allocation, and multicast configuration to ensure reliable, deterministic operation, particularly in mission-critical live environments.

Precise synchronization is achieved through IEEE 1588 Precision Time Protocol (PTP), which provides the sub-microsecond timing accuracy that broadcast facilities require. PTP not only keeps all devices locked to a common reference clock but also improves interoperability with audio-over-IP systems such as Dante, Ravenna, and Livewire.

One of the areas that has seen the most benefit from IP and ST 2110 is remote production. By maintaining video, audio, metadata, and control entirely within the IP domain, the need for dedicated SDI circuits between locations is eliminated. Modern telco data services can now carry these synchronized IP streams using standard high-capacity links. Although such connections remain technically demanding and cost-sensitive, they are far more accessible and no longer require telcos to maintain specialized broadcast infrastructure or expertise.

ST 2110 has become a cornerstone of the broadcast studio’s evolution, merging traditional engineering precision with the adaptability and scalability of IT networks, and paving the way for a new generation of flexible, connected, and distributed production workflows.