Deploying Large-Scale H 323 VoIP SP Networks

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Smooth migration to the Multimedia Generation Network

The most widely deployed use of H. SIP was designed to setup a "session" between two points and to be a modular, flexible component of the Internet architecture. It has a loose concept of a call that being a "session" with media streams , and has no intrinsic support for multipoint multimedia conferencing though implementers have built conferencing services to provide conferencing support. SIP is now. Though there have been some efforts to create a more modern communication standards, companies thus far have either elected to keep using these old systems, create proprietary systems that do not interwork to a significant degree, or focus effort on web-based conferencing i.

No communication system is simple, but H. SIP was initially focused on voice communication and then expanded to include video, application sharing, instant messaging, presence, etc. With each capability, complexity increases and, unfortunately, there are no strict guidelines as to what functionality any given device must support.

This leads to more complex systems with more interoperability problems. SIP was "marketed" as a simple protocol, in spite of the fact it only looks simple on the surface. Telephony is a hard problem and, regardless of how one wants to deliver it, the total system is going to have a certain level of complexity out of necessity. SIP has not defined procedures for handling device failure. If a proxy fails, the user agent detects this through timer expiration. It is the responsibility of the user-agent to send a re-INVITE to another proxy, leading to long delays in call establishment.

Messages are efficiently encoded and decoded by machines, with decoders widely available e. As a result, effort has been made to binary encode SIP e. Globally unique identifiers prevent feature and data element collision. SIP is extended by adding new header lines or message bodies that may be used by different vendors to serve different purposes, thus risking interoperability problems. The risk is admittedly small, but this problem has already been seen in the real world with similar extension schemes.

However, new revisions of H. SIP is extended by the standards community to add new features to SIP in such a way as to not impact existing features.

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However, new revisions of SIP are potentially not backward compatible e. In addition, several extensions are "mandatory" in some implementations, which cause interoperability problems. In addition, endpoints report their available and total capacity so that calls going to a set of gateways, for example, may be best distributed across those gateways.

There is no means of detecting the load on a particular gateway or to know whether a device has failed, meaning that proxies simply have to try a PSTN gateway, wait for the call to timeout, and then try another. When an H. In large networks, the direct call model may be used so that endpoints connect directly to one another. When using a SIP proxy to perform address resolution for the SIP device, the proxy is required to handle at least 3 full message exchanges for every call.

In large networks, such as IMS networks, the number of messages on the wire may be excessive. A basic call between two users may require as many as 30 messages on the wire! The H. The endpoint does not have to be concerned with the mechanics of this process, and the processing requirements for address resolution placed on the gatekeeper by H. Although out of scope of H. The endpoint does not have to be concerned with the mechanics of this process. Unfortunately, the processing requirements placed on the SIP proxy are higher than with H.

Flexible addressing mechanisms, including URIs, e-mail addresses, and E. Even with H. If the SIP proxy wants to collect billing information, it has no choice but to stay in the call signaling path for the entire duration of the call so that it can detect when the call completes. Even then, the statistics are skewed because the call signaling may have been delayed. Of course, more elaborate call establishment procedures may be required to negotiate complex capabilities, negotiate complex video modes, etc. Most real-world flows are more complex, as they often pass through one or more proxy devices, have intermediary response messages, and "negotiate" capabilities through a "trial and error" process that is far from scientific.

Here is a more real-life SIP call flow. Individual channels may be opened and closed during the call without disrupting the other channels. SIP entities have limited means of exchanging capabilities. RFC is the state of the art, which is more or less a "declaration" mechanism, not a negotiation procedure. The end result is still a "trial and error" approach in case the called party does not support the proposed media. SIP proxies can control the call signaling and may fork the call to any number of devices simultaneously. However, H.

How Media Gateway Controllers fit into the overall H. Pick the architecture you like best, I suppose. In addition, services may be provided to an endpoint as it places a call, as a call arrives, or during the middle of a call by a gatekeeper or other entity that routes the call signaling. As a result, H. SIP devices can receive service from a SIP proxy as the endpoint places a call, as a call arrives, or during the middle of a call. There is no defined way within SIP of providing services via a web browser or a feature server, as everything is done within the context of a "session".

However, there are no standards for this. Procedures are in place to provide control for the conference as well as lip synchronization of audio and video streams. SIP has limited support for video and no support for data conferencing protocols like T. SIP has no protocol to control the conference and there is no mechanism within SIP for lip synchronization.

There is no standard means of recovering from packet loss in a video stream to parallel H. A call can be made directly between two endpoints. However, most devices do utilize a gatekeeper for the purpose of registration and address resolution. SIP does not require a proxy. A call can be made directly between two user agents. However, most devices do utilize a SIP proxy for the purpose of registration, address resolution, and call routing.

No registration authority is required to use any codec in H. The SIP invite call setup message is time-sensitive, as the originator retries the call as quickly as possible if it does not proceed. This is useful in conditions where the SIP invite may be redirected through a number of servers before reaching the controller. This is a useful option when the SIP invite is directed through many servers before reaching the controller.

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This feature is disabled by default. When the disconnect extra call feature is enabled, the system monitors the number of active voice calls, and if the defined threshold is reached, any new calls are disconnected. The AP denies association requests from a device that is on call. You also need to enable call admission control in this profile. If you enable this feature, you can also configure the time duration within which the station should start the voice call after sending the TSPEC request the default is one second. Select the maximum time, in seconds, for the station to start the call after the.

Click Apply to save your settings. WMM works with WMM supports four access categories ACs : voice, video, best effort, and background. The When enabled, clients trigger the delivery of buffered data from the AP by sending a data frame. Select the SSID profile. In the Profile Details, select the Advanced tab. Select check this option. The IEEE DSCP classifies packets based on network policies and rules, not priority.

The PHB defines the policy and service applied to a packet when traversing the network. You configure these services in accordance with your network policies. All packets received are matched against the entries in the mapping table and prioritized accordingly. The mapping table contains information for upstream client to AP and downstream AP to client traffic.

Default mappings exist for all SSIDs. If you do not define a mapping for a particular DSCP tagged packet, default mappings are applied and prioritized accordingly Best Effort uses 0x When planning your mappings, make sure that any immediate switch or router does not have conflicting If this occurs, your traffic may not be prioritized correctly. Modify the DSCP mapping settings, as needed:. Traditional wireless networks provide all clients with equal bandwidth access. However, delays or reductions in throughput can adversely affect voice and video applications, resulting in disrupted VoIP conversations or dropped frames in a streamed video.

Thus, data streams that require strict latency and throughput need to be assigned higher traffic priority than other traffic types.

H versus SIP: A Comparison

This is defined as per the IEEE EDCA defines four access categories ACs to prioritize traffic: voice, video, best effort, and background. These ACs correspond to Traffic from legacy devices or traffic from applications or devices that do not support QoS.

For example, you can choose to give video traffic the highest priority. In the client, the data packets are then added to one of the transmit queues for voice, video, best effort, or background. The collision resolution algorithm responsible for traffic prioritization depends on the following configurable parameters for each AC:. The CW is reset to the minimum value after successful transmission. There are two sets of EDCA profiles you can configure:. Possible values are Divide the desired transmission duration by 32 to determine the value to configure.

With a value of 1 , the client reserves the access category through traffic specification TSPEC signaling.

H.323 Gatekeepers, Endpoints and Multipoint Control Units (MCU)

A value of 0 disables this option. WMM queue content enforcement is a firewall setting that you can enable to ensure that the voice priority is used for voice traffic. When this feature is enabled, if traffic to or from the user is inconsistent with the associated QoS policy for voice, the traffic is reclassified to best effort and data path counters incremented. Extended Voice and Video Functionalities. This section describes the other voice and video-related functionalities that are available on the controller. Voice and video devices use a signaling protocol to establish, control, and terminate voice and video calls.

These control or signaling sessions are usually permitted using pre-defined ACLs. When media traffic starts flowing, audio and video data are sent through that same port using RTP. The audio and video packets are interleaved in the air, though individual the sessions can be uniquely identified using their payload type and sequence numbers. Facetime users need to be assigned a role where traffic is allowed on these ports. WPA Fast Handover.

In the Check with the manufacturer of your handset to see if this feature is supported. This feature supports WPA clients, while opportunistic key caching also configured in the Select AAA profile. Select the When you enable IP mobility in a mobility domain, the proxy mobile IP module determines the home agent for a roaming client. ARM scanning on an AP during a call affects the voice quality. Select a profile instance from the drop-down menu to edit that profile.

Voice-Aware The Voice-Aware Although reauthentication and rekey timers are configurable on a per-SSID basis, an If a client is on a call, Using the WebUI to disable voice awareness for If you select AP Specific, select the name of the AP for which you want to disable voice awareness for Scroll down and deselect the Disable rekey and reauthentication for clients on call check box. Using the CLI to disable voice awareness for SIP Authentication Tracking. Upon successful registration, a user role is assigned to the SIP client. Select the AAA profile. Enter the configured user role for SIP authentication role.

Use the show voice client-status command to view the state of the client registration. Enable Real Time call quality analysis for the voice calls by selecting the Real-Time Analysis of voice calls check box. To view the detailed Real Time analysis report of a specific client, select the client and click the View Details button.

SIP session timer defines a keep alive mechanism for the SIP sessions using the periodic session refresh requests from the user agents.

Veraz Softswitch — ControlSwitch

The interval for the session refresh requests is determined through a negotiation mechanism. If a session refresh request is not received within the negotiated interval, the session is assumed to be terminated. For more information on the SIP session timer support, See section 8. Therefore, the ALG will not generate the responses for the session refresh requests. The range is - seconds. The default value is seconds. The Voice and Video Traffic Awareness for Encrypted Signaling Protocols support enables deep inspection of the traffic established over a secure layer to identify the voice or video sessions.

What is SIP?

In our example, we will configure this support for Microsoft Office Communicator. Click the Policies tab.

Select the Classify Media check box. Click Apply to apply the settings and save the configurations. Voice clients in an infrastructure can be switched to an alternate carrier or connection when they leave their active Wi-Fi coverage or roam to an area with poor Wi-Fi coverage. This process ensures QoS for voice calls. If the best signal strength reported by a voice client is equal to or less than the threshold value, the handover process is initiated. Expand Select the default profile. To configure the handover process do the following:. Select the Enable Handover Trigger feature checkbox.

The handover threshold value should be within the range 20 to 70 dbm. The default threshold value is dbm. Click the Apply button to save the configuration.

Eric A. Hall

The following command enables the dot11k profile and sets the handover threshold at dbm. The handover threshold value is a negative dbm value. In the CLI, enter the value without the negative - sign. After the dial plan is configured, a user can make SIP calls by dialing the destination number without any prefixes. The sequence number positions the dial plan in the list of dial plans configured in the controller. Examples of prefix codes are:. When the user dials a four digit number, no action is taken and the call is allowed. When the user dials a seven digit number, a nine 9 is prefixed to that number and the call is executed.

Example, if the user dials , the call is executed by adding 9 to the number, You can configure a maximum of two dial plan profiles and maximum of 20 dial plans per profile. Create a voice dial plan. Enter a name for the dial plan profile and click the Add button. Enter the following dial plan details and click the Add button. Enhanced Support. For information on call-handling, caller identification and callback capability, see the RedSky documentation.

The controller notifies the location of a voice client to the emergency server:. When a voice client roams from one access point to another access point in the same controller. When a voice client roams from one access point to another access point in a different controller.

The notification process ensures that the emergency call server is notified whenever a voice client is identified or the location of the client is updated. This may happen when there is a sudden loss of WLAN coverage due to extreme conditions such as, fire accidents. In such cases, the last associated access point will be the location of the voice client.

To track the location of a remote voice client, the administrator must configure the location of the remote access point in the controller or emergency call server. Voice over Remote Access Point. Voice traffic support is enhanced on split tunnel mode over a remote access point.

The voice traffic management for remote and local users are done on the controller. However, the sessions are created differently for both users. For remote users, the sessions are created on the remote access point and for local users, the sessions are created on the controller. This enhancement provides the following support for the voice traffic in the split tunnel over remote access point:. All voice ALGs work reliably in split tunnel mode when the PBX traffic is destined to flow through the corporate network.

Provides voice statistics and counters for remote voice clients in the split tunnel mode. Battery boost is an optional feature that can be enabled for any SSIDs that support voice traffic. This feature converts all broadcast and multicast traffic to unicast before delivery to the client. Enabling battery boost on an SSID allows you to set the DTIM interval from 10 - the previous allowed values were 1 or 2 , equating to 1, - 10, milliseconds.

This longer interval keeps associated wireless clients from activating their radios for multicast indication and delivery, leaving them in power-save mode longer, and thus lengthening battery life. An associated parameter available on some clients is the Listening Interval LI.

Benefits of Packet Telephony Networks

This defines the interval in number of beacons after which the client must wake to read the Traffic Indication Map TIM. The TIM indicates whether there is buffered unicast traffic for each sleeping client. With battery boost enabled, the DTIM is increased but multicast traffic is buffered and delivered as unicast. Increasing the LI can further increase battery life, but can also decrease client responsiveness. Click the Advanced tab.

Deploying Large-Scale H 323 VoIP SP Networks Deploying Large-Scale H 323 VoIP SP Networks
Deploying Large-Scale H 323 VoIP SP Networks Deploying Large-Scale H 323 VoIP SP Networks
Deploying Large-Scale H 323 VoIP SP Networks Deploying Large-Scale H 323 VoIP SP Networks
Deploying Large-Scale H 323 VoIP SP Networks Deploying Large-Scale H 323 VoIP SP Networks
Deploying Large-Scale H 323 VoIP SP Networks Deploying Large-Scale H 323 VoIP SP Networks

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