How To Deliver High Quality Telephony Over A Network

Transcription

1 Voice over Application Note Telephony Service over Satellite January 2012 Data Sells but Voice Pays In the early years of the industry, networks were deployed primarily for telephony services. As time went by, the trend became more data oriented. Today, almost every new network is all-ip delivering mainly data. But although voice/data ratios have shifted, the need for telephony services still exists. In view of the fact that consumers pay more for a bit of voice than they do for a bit of data, service providers can integrate telephony as part of their service portfolio in order to increase revenues. This application note demonstrates how service providers can deliver high quality telephony services over a network. 1 Gilat Satellite Networks Jan-12

2 Transparent VoIP or Full Telephony Service? A service provider can deliver voice services to customers over the network in a number of different ways. The solution will vary depending on the customer needs. The following are a few scenarios that service providers implement. Scenario 1 Transparent VoIP Hub Customer ATA or VoIP phone Customer softswitch In this scenario the service provider enables end-to-end VoIP connectivity but does not provide the VoIP equipment. At the side, the customer connects the VoIP phones or VoIP adapters to the LAN. At the hub side customers can use their own VoIP softswitch or connect to a 3 rd party VoIP service provider. The service provider is providing the customer with a transport network to transmit and receive VoIP sessions with the necessary QoS to assure call quality. This scenario is relevant when the customer is a large organization (e.g. an Enterprise) that manages their own VoIP system. 2 Gilat Satellite Networks Jan-12

3 Scenario 2 Integrated VoIP Solution Hub Analog phone Customer softswitch In this scenario, the service provider provides the customer with s that have embedded analog voice ports (FXS) for connecting a phone directly to the, but at the Hub site they provide a LAN interface for the customer to connect a softswitch. This scenario is relevant when the customer is a large organization with their own VoIP system that wants to use an integrated managed VoIP solution at the remote sites supporting standard analog phones. Scenario 3 Full Telephony Service Hub Softswitch PSTN In this scenario the service provider offers the customer an end-to-end telephony service including dialup lines and connectivity to the PSTN. At the side, the service provider delivers analog (FXS) ports on the router and/or enables support for IP phones. At the hub side, the service provider provides connectivity to the PSTN using its own softswitch or by connecting to a 3 rd party VoIP service provider. This scenario is relevant for customers who do not own/operate a voice switching system and require a full telephony service. In all three of these scenarios, the network needs to support advanced QoS and bandwidth management mechanisms for ensuring high voice quality. The following sections explain these mechanisms and demonstrate how they are implemented on Gilat s SkyEdge II platform. 3 Gilat Satellite Networks Jan-12

4 Guaranteed Voice over IP G-VoIP Guaranteed Voice over IP, or G-VoIP, is a set of features that make it possible for service providers to deliver high quality telephony services over satellite. The mechanisms described below are the building blocks of G-VoIP. Dedicated and Guaranteed Connection per Voice Call Each voice call carried over a network is transmitted as a single stream over a dedicated connection. The necessary bandwidth is allocated for each voice session by the in order maximize voice quality, minimize jitter and reduce delay. The system needs to ensure that there is no competition of resources between VoIP calls, no competition of resources with other traffic, and no competition of resources with other s. This is achieved by creating a dedicated tunnel between the two s or the and the hub that is established once the call is initiated and maintained throughout the entire call. Figure 1 - Dedicated and Guaranteed Connection per Voice Call 4 Gilat Satellite Networks Jan-12

5 Call Admission Control In order to ensure voice quality in a telephony service, it is necessary to have sufficient bandwidth resources at all times. A new VoIP call without enough resources will lead to high jitter and packet loss resulting in poor voice quality and will also affect all existing calls. The solution to this is one of the G-VoIP mechanisms implemented in the SkyEdge II system called Call Admission Control (CAC). For each, new VoIP calls will be admitted to the network only if there are enough bandwidth resources network wide to comply to the QoS policy and service level for that. If there are not enough bandwidth resources, the call will be not admitted and a signal to generate a busy tone will be send back to the softswitch to prevent network congestion. With CAC, there is no compromise on voice quality for existing calls. Figure 2 - Call Admission Control 5 Gilat Satellite Networks Jan-12

6 Enhanced Inbound Bandwidth Reservation With the typical high round trip delay in networks (over 500ms), it is critical to minimize the response time for adaptimg the inbound space segment resources to address the changing needs of the network without compromising bandwidth efficiency. SkyEdge II addresses this challenge by creating a new bandwidth allocation plan every 40ms (i.e. 25 times a second). With this highly frequent reevaluation of network needs, delay sensitive applications are addressed as quickly as possible. In the example shown in Figure 3, three s are allocated timeslots over two 40ms windows. Figure 3: Timeslot allocation example A is provided with a guaranteed rate and a fixed timeslot for the VoIP traffic to prevent jitter. s B and C use a combination of rate and volume based allocations for the HTTP and FTP traffic. Since this traffic is not sensitive to jitter the timeslot position may vary. C receives a larger allocation to support the higher volume of the FTP upload. 6 Gilat Satellite Networks Jan-12

7 Advanced Outbound Scheduling In SkyEdge II the outbound packet scheduling mechanism favors real time applications such as VoIP. VoIP packets are sent immediately for transmission while other low priority packets are delayed a bit for the next frame with a suitable ModCod. This algorithm is uniquely implemented in the Gilat IP Modulator, a component of the SkyEdge II hub. In the following example, the adaptive nature of the frames in DVB-S2 can result in scheduling conflicts. Consider the case presented in Figure 4. Receiving can tolerate 8PSK 7/8 VoIP #134 Internet #768 Receiving limited to QPSK 3/4 Frame 765 #490 VoIP #134 VoIP #134 Frame 766 QPSK, 3/4 8PSK, 7/8? Figure 4: Real time scheduling The VoIP packet would normally be scheduled for transmission in the next frame but ideally it should be sent immediately to reduce jitter. The modulator will favor the high priority VoIP packet and will scheduled it for immediate transmission while the Internet data packet will be delayed for the next suitable frame. As with the other mechanisms described here, this reduces jitter and improves the user call experience. G-VoIP QoS and SLA A G-VoIP enabled such as SkyEdge II can be configured with maximal VoIP bit rate VoIP MIR. By using this feature, a can be limited with the maximal number of simultaneous calls/sessions. For example, SIP video call will have 2 sessions, 1 for Voice and another for Video. This can be used by the service provider to create services with different pricing levels. 7 Gilat Satellite Networks Jan-12

8 C-RTP Compression In order to ensure the most efficient space segment utilization SkyEdge II implements RTP header compression technique that compresses the RTP frame header by 95% from 40 Bytes to 2 Bytes. Since VoIP payload is short, this technique reduces the effective VoIP bit rate by as much as 75% allowing supporting up to 4 times VoIP calls on the same bandwidth of one VoIP call without compression. Figure 5: Inbound header compression Star and Mesh Topologies SkyEdge II supports both star and mesh topologies simultaneously in a single network. Calls may be established between a SkyEdge II terminal and the hub, or single-hop between meshenabled s improving voice quality and significantly cutting down on delay and bandwidth usage. Mesh Star Star SkyEdge II Pro Embedded ATA SkyEdge II Hub SkyEdge II Access Embedded ATA Analog or VoIP phone Softswitch SkyEdge II IP ATA SkyEdge II Access Embedded ATA PSTN High Speed Internet Figure 6: Star and Mesh Topology 8 Gilat Satellite Networks Jan-12

9 Analog Phone Ports on the In order to support analog phones at the remote sites, a needs to have the option for FXS ports. The SkyEdge II product portfolio includes models that can accommodate add-on ATA cards with FXS ports that can be inserted into the. These ATA cards have 2 phone ports thereby providing up to 4 embedded telephony ports on the SkyEdge II Access or 8 ports on the SkyEdge II Pro. The embedded ATA card is managed and monitored by Gilat s NMS allowing remote configuration and monitoring. IP Phones may also be directly connected to the embedded LAN ports on any of the SkyEdge II s. SkyEdge II Pro with 2 x FXS card (4 voice ports) and Mesh receiver card SkyEdge II Access with 2 x FXS card (4 voice ports) Figure 7: SkyEdge II Access and Pro with FXS add-on cards Summary Adding voice services is a way for service providers to increase their revenues. Gilat has maintained a dominant market position in the voice over market and has implemented its knowledge and expertise in this field in the architecture of SkyEdge II. The G-VoIP features mentioned in this application note enable service providers to offer differentiated voice services that meet the stringent SLA requirements for delivering a high quality voice service. For more information, please visit 9 Gilat Satellite Networks Jan-12