When the number of hosts increases 11 to 20 , a shared medium spanning over the whole LAN cannot cope with the real-time traffic. This is due to the fact that, in a shared medium, the number of frame collisions at the physical layer increases exponentially with the number of transmitted frames. Delay and jitter in frame transmission over the network is proportional to the number of frame collisions. Therefore delay and jitter in delivery of real-time traffic increases exponentially with the number of frames transmitted.
Please see 3. The following actions may be taken to improve the QoS offered by the HO's local network to multimedia communication:. Also HO routers might support QoS mechanisms, that allow expedited forwarding of voice and video packets to the WAN before the packets belonging to other less prioritary applications like HTTP connections.
The QoS mechanisms help to decrease the delay produced by long-duration, high-bandwidth data bursts, as well as delay variation, or jitter, produced by transmission of packets of different lengths over the link to the WAN. If an HO router is present, it is recommended that it supports the same QoS mechanisms as multimedia terminals and computers in the HO.
Video should be disabled in HOs using low speed modems such as V. Video should also be disabled in HOs that must support simultaneous voice and data communication or several voice communications at a time but are not using a high-speed modem such as an xDSL modem.
For an SO that connects to the main network through a modem, please see 3. This is because the higher the number of hosts is, the more traffic local to the LAN, i.
There are two methods to segment a LAN: bridging or switching. Bridging is cheaper and simpler, but it decreases LAN congestion in a minor way compared to switching. Switching is the most expensive solution, but it can simply dissolve the LAN congestion problems if the number of available switch ports is high enough. Notice that increasing the bandwidth to the WAN, either by adding a new point-to-point link or by leasing more capacity on an existing link, will affect the BO router.
This is because the higher the number of hosts, the more traffic local to the LAN, i. A normal LAN is kept below congestion conditions by correctly dimensioned capacities and layout of transmission media and LAN devices like hubs, bridges and L2 switches. In a normal LAN every frame is equally important, and all of them are dealt with in the same way. However, when adding multimedia terminals to a normal LAN the amount of traffic running over this LAN is greatly increased.
Besides, frames carrying voice packets demand a much more expedited treatment from the LAN than frames carrying e. From the above it can be concluded that the following measures may help to improve the QoS offered by the LAN to multimedia traffic:. Also if the LAN is running on half-duplex transmission media, it is recommended to migrate to full-duplex. That really doubles the available bandwidth for every host in the LAN. Applications are only aware of Layer 3 QoS mechanisms.
Please refer to the documentation accompanying the application to find out what Layer 3 QoS mechanisms that are supported. IEEE Please refer to the documentation accompanying the devices to find out what Layer 2 QoS mechanisms that are supported. Please refer to the documentation accompanying the devices to find out what mappings that are supported.
If this is not the case, segmentation of the LAN is necessary. There are two ways to segment a LAN. They are thoroughly described below. If the action proposed in point 1 above results too expensive, or as a complement to that action, it is recommendable to decrease the number of users of every LAN segment to a minimum ideally one.
To see how a bridged LAN looks, see Figure Bridges are half-intelligent devices, in the sense that they only forward a frame to one end of the bridge if the destination for that frame is not on the opposite end except for broadcast frames, as will be seen in next paragraph.
However, bridges do not stop broadcast frames from spanning over all the segments in the bridged LAN. This means that broadcast storms like ARP storms described before are not affected by this method. Neither does this method stop frames from traversing several shared LAN segments until they get to their destinations. Therefore collisions may take place in one shared segment that are caused by frames from other segments.
This method does not require new cabling, only the bridges must be added to the LAN. Old hubs may be re-used. It should be investigated if a bridging solution is enough for the LAN before deciding for a more expensive solution, i. To see how a switched LAN looks, see Figure Since a Layer 2 switch is a concentrator device instead of an interconnection device like a bridge, frames between users on different LAN segments do traverse two LAN segments only: the originating one, and the destination one.
This limits the collision domain for a frame to maximum two shared LAN segments. Besides, switches allow LANs to be hierarchically segmented, by having for instance low-level switches to which small LAN segments are connected, in turn connected to a higher switching level. The ideal segmentation is that in which there is one switch port available for every user.
Moreover, if the transmission media used in a switched LAN are full-duplex, frame collision never takes place in the LAN , greatly improving frame delay and bandwidth.
A switched LAN 3. These links are point-to-point between devices. However sometimes several devices are interconnected using a small LAN for reliability reasons if one of them becomes inoperative, another one connected to the same LAN can be reached.
The following actions may be taken to improve the QoS offered by the WAN to multimedia communications:. Notice that for this measure to be effective the QoS mechanisms supported by the WAN devices must be the same as that supported by terminals and LAN devices. However it is possible to support a different QoS mechanism on another port in the same device, like the one described in the paragraph above.
Summarizing, it is required that all devices connected to a port in a WAN device support the same QoS mechanism as the port in question. These mechanisms may speed up the delivery of high-priority, low-delay packets; but they will do it at the cost of delaying the delivery of other packets even more.
Moreover, even though delivery of high-priority packets may be speeded up, if the WAN device is congested, it will probably not provide the QoS required for a real-time peer-to-peer communication.
If congestion is found in WAN devices, this action must be taken together with the two actions in points 1 and 2 above. From the previous section, it can be seen that all network areas are suitable to be improved for multimedia communication by enabling QoS mechanisms in them.
This is done by adding network devices and end systems with support for one or more QoS mechanisms. The goal of this process is to get an improved QoS performance from end to end over the whole network. This can be an expensive and lengthy process.
Besides, not all of the QoS mechanisms available are equally suited for all network areas. In the paragraphs below a guideline for how and when to deploy QoS mechanisms in a network is provided. It is obvious that to deploy QoS mechanisms over the whole IP network at the same time is very difficult. Deployment of these mechanisms is better done in steps. Finally, the most delicate deployment takes place in WAN backbones, mainly composed of L3 switches and high-end routers. For the WAN border, i.
In the WAN core, i. The choice is up to the network designer. For a summary of the paragraphs above, see Figure Suitability areas for QoS mechanisms Anyway, no matter which QoS mechanisms are chosen, to achieve end-to-end improved QoS for an existing network is a long and difficult task. In many cases it will imply replacing network devices, adding new modules and software to existing devices, and managing new network configuration items like queue lengths, queue priorities, classes of service etc.
The reward to all this effort is, of course, a unified network on which any kind of information can be carried without troubles. There are security issues affecting the ability of a network to support real-time multimedia communications. Firewalls, proxies and RAS systems may disallow this type of communications if not properly installed.
No Proxies or Firewalls should be in the path of multimedia communications. The ASB 04 system offers a feature that uses an IP network for real-time multimedia and data conferencing between extensions belonging to one ASB 04 system calls between extensions belonging to different ASB 04 system cannot be set up using the IP network.
Now communication between users is not limited to voice calls, but multimedia conferences with video and audio, document sharing, interactive whiteboard etc. The well-proven, full-featured ASB 04 system makes its rich set of supplementary services available to terminals connected to the IP network.
Only industry-standard protocols are used in the IP Extension feature. Here just a brief description is provided, with enough information as to make one able to plan and prepare the installation of the feature and terminals. Any H. It is important to remark that, even though they claim so, most of the terminals and clients in the market are not completely H.
Either some features are missing, or they implement misinterpretations of the standard. The IP Extension feature is a highly standard-compliant product. To invoke supplementary services, handling the terminal's display, present the terminal's user with options and other related functions, ASB 04 includes a built-in WAP server with full support of the WML standard. Any terminal compliant to H. The following terminals fulfil the requirements in the paragraph above, and are therefore recommended to use with the IP Extension feature.
The following devices are unable to request supplementary services to the IP Extension feature, but can be included as passive party in execution of any of these services like e. Finally, the following devices, even though not being completely standard, can be used to place basic calls, i. Despite the above list, any other device compliant to the basic H. Support for terminal-received as opposite to terminal-invoked supplementary services requires that the terminal fully complies to the H.
Consult the technical staff of the terminal's vendor to find out whether it complies with this standard procedure. Otherwise, execution of supplementary services like transfer, conference etc. In the same way, IP telephones connected to the Network are not different from the network's point of view from PCs or other end-user products. The following list highlights the relevant issues to bear in mind when deploying an ASB 04 system with an integrated IP Extension feature.
One good point of connection is to the LAN segment where the server farm is located. This segment is supposed to have high bandwidth and few machines using it. Of course, before adding up the new server the ASB 04 IP Extension feature one must analyze this segment to make sure that the added load due to multimedia communications can be supported and that the delay and jitter conditions of packets flowing through this LAN segment will be optimal.
If this is not the case, it will be necessary to upgrade the server farm LAN segment. Apart from the ASB 04 system and its IP Extension feature, there are some other considerations specific to certain network devices that must be taken into account when deploying multimedia communication over a packet network. Standard off-the-shelf H. It is important to remark that PC-based applications provide worse QoS than specialized terminals like e.
IP telephones. Specifically, lower voice quality and longer delays are obtained when using a PC-based application for real time communications. On some routers, you can allocate a portion of the available bandwidth specifically for voice traffic. How to Calculate Voice Bandwidth If it is a new installation, you may not be able to monitor the voice traffic to find out the bandwidth for the peak calls.
There is a way to calculate the number of channels using Erlang B tables. Erlangs are a unit of communication that will be used to look-up the number of channels from a Erlang B table. The table will indicate the number of channels required for a certain percentage of blocking. The percentage of blocking refers to how often an incoming call will be busy. Just a note, this blocking not only affects incoming calls but outgoing calls also.
The first step is to monitor or estimate the number of incoming and outging calls during peak hours. Next monitor or estimate the average time per call in seconds. For this example, the average call will last Now we run these two numbers through this formula to calculate the number of Erlangs: In case you were wondering, the number is the number of seconds in an hour and is used to make sure that the units hours and seconds cancel.
Next we need to determine the number of voice channels that are needed to meet the required blocking. For our example, we select the closest value to 8. That would require a minimum of 13 channels. Now we can determine the bandwidth required. Video will likewise use Kbps total to carry an upstream and downstream connection. To cope with unexpected spikes in traffic and increased usage over time, Skype for Business Server media endpoints can adapt to varying network conditions and support three times the throughput for audio and video while still maintaining acceptable quality.
Do not assume that this adaptability will mask the problem when a network is under-provisioned. In an under-provisioned network, the ability of the Skype for Business Server media endpoints to dynamically deal with varying network conditions for example, temporary high packet loss is reduced.
For network links where provisioning is very costly and difficult, you may have to consider provisioning for a lower volume of traffic. In this scenario, let the elasticity of the Skype for Business Server media endpoints absorb the difference between the traffic volume and the peak traffic level, at the cost of some reduction in the voice quality. Also, there will be a decrease in the headroom otherwise available to absorb sudden peaks in traffic. For links that cannot be provisioned correctly in the short term for example, a site that uses very poor WAN links , consider disabling video for certain users.
Provision the network to guarantee a maximum end-to-end delay latency of milliseconds ms under peak load. Latency is the one network impairment that Skype for Business Server media components can't reduce, and it is important to find and eliminate the weak points.
For servers that are running antivirus software, include all servers that are running Skype for Business Server in the exception list to provide optimal performance and audio quality. The recommendation is motivated by the need to avoid any delay in the allocation of media ports due to IPsec negotiation. The bandwidth used to download conference content from the Internet Information Services IIS server depends on the size of the content. You may choose to monitor the actual usage and adjust bandwidth planning accordingly.
An important part of network planning is ensuring that your network can handle the media traffic generated by Skype for Business Server. This section helps you plan for that media traffic. The media traffic bandwidth usage can be challenging to calculate because of the number of different variables, such as codec usage, resolution, and activity levels.
The bandwidth usage is a function of the codec that is used and the activity of the stream, which can vary between scenarios. The following table lists the audio codecs typically used in Skype for Business Server scenarios. If enough bandwidth is not available for that codec, then calls can fail with an error that resembles the following in the Media logs: Atleast one codec must be enabled, hr: c Media logs.
The bandwidth numbers in the previous table are based on 20ms packetization 50 packets per second and for the Siren and G.
Forward Error Correction FEC is used dynamically when there is packet loss on the link to help maintain the quality of the audio stream. The stereo version of the G. The default codec for video is the H. To maintain interoperability with legacy clients, the RTVideo codec is still used for peer-to-peer calls between Skype for Business Server and legacy clients.
In conference sessions with both Skype for Business Server and legacy clients the Skype for Business Server endpoint may encode the video using both video codecs and send the H. The bandwidth required depends on the resolution, quality, frame rate, and the amount of motion or change in the picture. For each resolution, there are two pertinent bit rates:. Maximum payload bit rate This is the bit rate that an endpoint will use for resolution at the maximum frame rate.
This is the value that allows the highest video and sound quality. Minimum payload bit rate This is the bit rate below which a Skype for Business Server endpoint will switch to the next lower resolution. To guarantee a certain resolution, the available video payload bit rate must not fall below this minimum bit rate for that resolution.
This value is helps you understand the lowest value possible if the maximum bit rate is not available or practical. For some users, such a low bit rate video might provide an unacceptable video experience so use caution with these minimum video payload bitrates.
Note that for static, unchanging video scenes the actual bit rate may temporarily fall below the minimum bit rate. Skype for Business Server supports many resolutions. This allows Skype for Business Server to adjust to different network bandwidth and receiving client capabilities.
The default aspect ratio for Skype for Business Server is The legacy aspect ratio is still supported for webcams which don't allow capture in the aspect ratio. Endpoints do not stream audio or video packets continuously. Depending on the scenario there are different levels of stream activity which indicate how often packets are sent for a stream. The activity of a stream depends on the media and the scenario, and does not depend on the codec being used.
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