Lab 3 - RTP Header Compression
Lab 3 - RTP Header Compression
net
The objective of this lab exercise is for you to learn and understand how implement Compressed RTP on
Cisco IOS H.323 gateways
Topic Description:
Compressed RTP (cRTP) reduces the RTP header to 2 or 4 bytes. It is used to save bandw idth on WAN
links in VoIP networks
Lab Difficulty:
Readiness Assessment:
When ready for your exam, you should be able to complete this lab in no more than 10 minutes
Lab Topology:
Configure the hostnames and IP addresses on R2 and R4 as illustrated in the diagram. Configure R4 to
provide clocking to R2. Configure the clock rate on R4 as 800Kbps
Task 2:
Configure POTS dial peers on R2 using the information in the diagram. You are permitted to use any dial
peer numbers that you want to use
Task 3:
Configure POTS dial peers on R4 using the information in the diagram. You are permitted to use any dial
peer numbers that you want to use
Task 4:
Configure a single VoIP dial peer on R2 to the R4 extension range. You are permitted to use any dial
peer numbers that you want to use. Use SIP as the signaling protocol for the dial peer
Task 5:
Configure a single VoIP dial peer on R4 to the R2 extension range. You are permitted to use any dial
peer numbers that you want to use. Use SIP as the signaling protocol for the dial peer
Task 6:
Task 7:
Task 8:
Verify your configuration using the appropriate Cisco IOS show commands
R2(config)#hostname R2
R2(config)#interface Serial0/0
R2(config-if)#description .Connected To R4 Serial 0/1.
R2(config-if)#ip address 150.1.1.2 255.255.255.0
R4(config)#hostname R4
R4(config)#interface Serial0/1
R4(config-if)#description .Connected To R2 Serial 0/0
R4(config-if)#ip address 150.1.1.4 255.255.255.0
R4(config-if)#clock rate 800000
Verify connectivity by performing a simple ping betw een the two routers
R2#ping 150.1.1.4
R4#ping 150.1.1.2
Task 2:
Task 3:
Task 3:
R4(config)#dial-peer voice 1 pots
R4(config-dial-peer)#destination-pattern 2211
R4(config-dial-peer)#port 3/0/0
R4(config-dial-peer)#exit
R4(config)#dial-peer voice 2 pots
R4(config-dial-peer)#destination-pattern 2221
R4(config-dial-peer)#port 3/0/1
R4(config-dial-peer)#exit
Task 4:
Task 5:
Task 6:
R2(config)#int s0/0
R2(config-if)#ip rtp header-compression
R2(config-if)#exit
Task 7:
R4(config)#int s0/1
R4(config-if)#ip rtp header-compression passive
R4(config-if)#exit
Task 8:
The first validation task is to place a call from a Site 1 extension to a Site 2 extension:
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 1111
Called Number : 2222
Bit Flags : 0x12120030 0x100000 0x400
CC Call ID :3
Source IP Address (Sig ): 150.1.1.2
Destn SIP Req Addr:Port : 150.1.1.4:5060
Destn SIP Resp Addr:Port: 150.1.1.4:5060
Destination Name : 150.1.1.4
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID :3
Stream Type : voice-only (0)
Negotiated Codec : g729r8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
Media Source IP Addr:Port: 150.1.1.2:18310
Media Dest IP Addr:Port : 150.1.1.4:16844
Orig Media Dest IP Addr:Port : 0.0.0.0:0
Next, verify that RTP traffic is being compressed. To generate RTP traffic, you must have physical access
to the phone and talk, etc, to send packets across the WAN. If you do not have physical access to the
phone, simply use this as a command verification exercise
R2#show running-config
Building configuration...
R4#show running-config
Building configuration...