Alvarion SIP R2J, Voice Gateway

System Menu

Voice Gateways System Manual 67

3.9.5 RTP Statistics Page

The RTP Stats page enables to monitor from remote the performance of the last

call. The displayed information includes: Bandwidth (kb/s), jitter, packet loss,

and latency.

NOTE

The bandwidth relates to the payload only.

Figure 3-25: RTP Statistics Page

Contents

Main Document History Page Legal Rights Trade Names Statement of Conditions Warranties and Disclaimers Exclusive Warranty Disclaimer Limitation of Liability Disposal of Electronic and Electrical Waste Important Notice About This Manual Contents Chapter 1 - System Description Chapter 2 - Installation Chapter 3 - Using the Web Configuration Server Appendix A - Internal Class 5 Services Appendix B - Default Telephony Parameters Page Page 1.1 Introducing the Voice Gateway 1.2 Specifications 1.2.1 Telephony and Fax Services 1.2.2 Security 1.2.3 Voice Quality 1.2.4 Configuration and Management 1.2.5 Bridge Functionality 1.2.6 Mechanical 1.2.7 Electrical 1.2.8 Connectors 1.2.9 Regulatory Standards Compliance 1.2.10 Environmental Page 2.1 Installation Requirements 2.1.1 Packing List 2.1.2 Additional Installation Requirements 2.2 Front and Rear Panel Components 2.2.1 Connectors 2.2.2 Reset to Factory Default Configuration 2.2.3 LEDs 2.3 Installation and Commissioning Page Page 2.4 Notes on Using the Voice Gateways in Alvarion's Systems 2.4.1 BreezeMAX System (Version 1.5 and higher) 2.4.2 BreezeACCESS VL System (Version 3.1) Chapter 3 - Using the Web Configuration Server In This Chapter: Page 3.2 Accessing the Web Configuration Server 3.3 Using the Web Configuration Server Page 3.4 Home Menu - Product Info Page Page 3.5 WAN Menu 3.5.1 WAN Status Page Page 3.5.2 WAN Configuration Page Page 3.6 VLAN Tagging Menu 3.6.1 VLAN Tagging Page 3.6.2 Adding and Deleting VLANs Page Page 3.6.4 VLAN Configuration Example 1 Untagged POTS 3.6.5 VLAN Configuration Example 2 3.6.5.1 VG-1 Configuration 3.6.5.2 VG-2 Configuration 3.6.6 VLAN Configuration Example 3 3.6.6.1 Method 1 3.6.6.2 Method 2 Page 3.7 Telephone Menu Page Page Page Page Page Page The jitter buffer options are common to both lines (if applicable): The following settings are available for each line: Page The SIP Extensions page includes the following components: 3.7.2 SIP Extensions Page Page 3.7.3 NAT Traversal Configuration Page (SIP Only) 3.7.4 STUN Client Configuration Page (SIP only) 3.7.5 ToS Page 3.7.6 Line Configuration Page 3.7.7 Dial Plan Schemes Page 3.7.7.1 Hotline 3.7.7.2 Adding/Removing Prefixes Page Page Page Page 3.9 System Menu 3.9.1 Set Security Password Page Page 3.9.2 Localization Page The Localization page includes the following components: 3.9.3 SNMP Configuration Page The SNMP Configuration page includes the following components: 3.9.4 Service Access Configuration Page Page 3.10 Upgrade Page 3.10.1 Downloader Result Codes (hexadecimal) 3.10.1.1 Examples Page Page 3.13 Parameters Summary Page Page Page Page Page A.1 Actions and Keypad Sequences A.2 Using the Class 5 Services A.2.1 Call Waiting A.2.2 Call Inquiry A.2.3 Call Alteration A.2.4 Call Drop A.2.5 3-Party Conference 1 Description: One on-going call active, place a new call to a third party and start conference. Description: Available only when there are no calls active/in progress. A.2.6 3-Party Conference 2 Description: Two on-going calls active, mix them into a conference session. A.2.8 Call Forward Description: Available only when there are no calls active/in progress. Page Page Page Page Page Page C.1 Metering Support C.2 Sending VoIP Performance Data to a Remote System Page Page Glossary