Avaya 2400 Service Guide

Avaya Solution & Interoperability Test Lab

Application Notes for Configuring compressed Real Time Protocol over Multi-Link Point-to-Point Protocol between Juniper Networks J4300 and M7i routers to Support an Avaya IP Telephony Infrastructure – Issue 1.0

Abstract

These Application Notes describe the steps for configuring Juniper Networks J4300 and M7i routers to use compressed RTP (cRTP) over a Multi-Link Point-to-Point Protocol (MLPPP) connection to support an Avaya IP Telephony Infrastructure consisting of Avaya Communication Manager and Avaya IP Telephones. The Juniper Networks routers will perform header compression for all RTP traffic traversing over the MLPPP connection to minimize overhead used by the RTP packets thus increasing available bandwidth, and load distribution across the multi-link bundle for increase bandwidth and resiliency.

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1. Introduction

Real Time Protocol (RTP) packets generated by Voice over IP (VoIP) telephony are typically small in size ranging in tens of bytes per packet. IP (20 bytes) and UDP (8 bytes) headers are then added onto each packet before transmission. Because of the relative small packet size of RTP packet, the IP and UDP headers are all overhead. For RTP packets that traverses a Wide Area Network (WAN) with limited bandwidth, these headers represents an opportunity for bandwidth saving that could otherwise be use for other traffic or additional VoIP calls. This is the main idea behind the use of cRTP.

It is most common to use the G.729 codec for calls across a low speed link due to its lower bandwidth requirement, but either G.711 or G.729 codecs can benefit from cRTP. Both G.711 and G.729 codec were exercised during compliance testing.

In addition, the Juniper J4300 and M7i router also has the ability to distribute VoIP traffic across all members on a per flow basis. The Juniper routers accomplished per flow load distribution through the examination of the 5tuples (Source/Destination IP, Source/Destination Port, and protocol) in each packet. Since these 5 pieces of information are the same for any given call flow, all rtp packets for a particular call will always be distributed onto the same multilink member. This is an important feature as members within a multilink bundle may be from different Service Provider and the links may have varying delay characteristic. Therefore, the ability of performing per flow load balancing can help minimize jitter in VoIP application.

Figure 1, shows the sample network used in these Application Notes. Two separate IP networks, one in each location are connected together by a pair of Juniper Networks routers over a MLPPP connection. Each location contains an Avaya Media Server, an Avaya Media Gateway, and Avaya IP Telephones. A dial plan and an H.323 trunk configured between the two Avaya Communication Managers allow calls to be routed between the two systems. Both the Juniper Networks M7i and J4300 routers are configured to perform RTP header compression for all RTP packets traversing over the PPP connection. Both routers are configured to prioritize VoIP traffic based on DiffServ Code Point (DSCP) information encoded in each VoIP packet. Bandwidth allocation was set on all interfaces shown to guarantee necessary bandwidth is reserved for VoIP traffic in the event of network congestion. Both Juniper routers will statistically distribute the call flows across all members of the MLPPP bundle. The combination of these elements provides the necessary Quality of Service for VoIP traffic traversing over the WAN connection.

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2. Configuration

Figure 1 illustrates the configuration used in these Application Notes. Telephones with range number 3xxxx are registered with the Avaya S8300 Media Server on the right side of the figure, and telephones with extension range 2xxxx are registered with the Avaya S8500 Media Server on the left side of the figure. An H.323 IP trunk was used to route calls between the two Avaya Media Servers. Note that extensions from both Avaya Communication Manager systems are located in each location. This is done to verify the Avaya IP Telephones can register and place call successfully through a cRTP enabled WAN connection.

AL; Reviewed: SPOC 2/12/2007

Figure 1: Sample Network Configuration

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3. Equipment and Software Validated

The following equipment and software/firmware were used for the sample configuration:

Equipment Software/Firmware

Avaya S8300 Media Server with G350 Media Gateway Avaya S8500 Media Server Avaya Communication Manager R3.1.2

Avaya G650 Media Gateway -

TN2312BP IPSI HW03 FW 22 TN799DP C-LAN HW01 FW 16

TN2302AP IP MedPro HW18 FW 108 Analog telephone N/A Avaya 6408D digital Telephone N/A Avaya 4602SW IP Telephone (H.323) R2.3 – Application (a10d01b2_3.bin) Avaya 4610SW IP Telephone (H.323) R2.6 – Application (a10d01b2_6.bin) Avaya 4620SW IP Telephone (H.323) R.2.6 – Application (a20d01b2_6.bin) Avaya IP Softphone R5.24.8 Juniper Networks J4300 router JUNOS 8.1R1.5 Juniper Networks M7i router JUNOS 8.1R1.5

Avaya Communication Manager R3.1.2

(R013x.01.2.632.1)

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4. Avaya Communication Manager

There is no unique configuration required in Avaya Communication Manager to support compressed RTP (cRTP) or any feature mentioned in this document. For detailed information on the Installation, Maintenance, and Configuration of Avaya Communication Manager, please consult reference [1], [2], and [3].

Step Description

Below is the output from the display ip-network-region command showing the MEDIA PARAMETERS, and DIFFSERV/TOS PARAMETERS information

configured in Avaya Communication Manager. All traffic used in the sample network is configured for network region 1.

The Call Control PHB Value of 34 is equivalent to 100010 in binary. The Audio PHB Value of 46 is equivalent to 101110 in binary.

The MEDIA PARAMETERS, and DIFFSERV/TOS PARAMETERS information will be needed in later sections when configuring the routers.

display ip-network-region 1 Page 1 of 19 IP NETWORK REGION Region: 1 Location: Authoritative Domain: Name: MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 1 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? n

UDP Port Max: 3029 DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y Call Control PHB Value: 34 RTCP MONITOR SERVER PARAMETERS Audio PHB Value: 46 Use Default Server Parameters? y

Video PHB Value: 26

802.1P/Q PARAMETERS Call Control 802.1p Priority: 6 Audio 802.1p Priority: 6 Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS H.323 IP ENDPOINTS RSVP Enabled? n H.323 Link Bounce Recovery? y Idle Traffic Interval (sec): 20 Keep-Alive Interval (sec): 5

Keep-Alive Count: 5

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5. Configure the Juniper Networks Routers

The following sections describe the steps for configuring the different Juniper Networks routers in the sample configuration. Unless otherwise specified, all router configurations are based on Juniper Networks recommendation.

5.1. Configure the Juniper Networks J4300 Router

This section shows the necessary steps in configuring the Juniper J4300 router as shown in the sample network. The following steps use the Command Line Interface (CLI) offered by the J4300 router.

Step Description

Connect to the J4300. Log in using the appropriate Login ID and Password.

The following prompt will appears after successful log in.

interop@J4300>

Enter configuration mode by typing in edit at the prompt.

interop@J4300> edit interop@J4300#

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Step Description

Configure the code-point-aliases and classifier for Avaya VoIP traffic.

• The alias helps identify the binary dscp setting.

• The sample network uses the name “avaya-rtp” to denote dscp binary bit 101110

for media traffic. This is equivalent to the decimal Audio PHB Value of 46 set in Avaya Communication Manager for RTP Media in Section 4, Step 1.

• The sample network uses the name “avaya-sig” to denote dscp binary bit 100010

for signaling traffic. This is equivalent to the decimal Call Control PHB Value of 34 set in Avaya Communication Manager for signaling in Section 4, Step 1.

interop@J4300# edit class-of-service code-point-aliases interop@J4300# set dscp avaya-rtp 101110 interop@J4300# set dscp avaya-sig 100010 interop@J4300# exit

• Define a classifier called “Avaya-voip”.

• The classifier “Avaya-voip” defines the forwarding characteristic used by the

router based on traffic type.

• The sample configuration is configured to use expedited-forwarding with low

loss-priority for “avaya-rtp”, and assured-forwarding with low loss-priority for “avaya-sig” to ensure proper Quality of Service (QoS) priority is assigned to voice traffic.

interop@J4300# edit class-of-service classifiers interop@J4300# edit dscp Avaya-voip interop@J4300# set forwarding-class expedited-forwarding loss-priority

low code-points avaya-rtp

interop@J4300# set forwarding-class assured-forwarding loss-priority

low code-points avaya-sig

interop@J4300# exit

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