Cisco Systems PAP2T, SPA2102, SPA3102, WRP400, SPA8000 manual Click Voice tab System

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Configuring Your System for ITSP Interoperability

3

 

Network Address Translation (NAT) and Voice over IP (VoIP)

 

 

 

 

 

Determining Whether the Router Uses Symmetric or

Asymmetric NAT

STUN does not work on routers with symmetric NAT. With symmetric NAT, IP addresses are mapped from one internal IP address and port to one external, routable destination IP address and port. If another packet is sent from the same source IP address and port to a different destination, then a different IP address and port number combination is used. This method is restrictive because an external host can send a packet to a particular port on the internal host only if the internal host first sent a packet from that port to the external host.

NOTE This procedure assumes that a syslog server is configured and is ready to receive syslog messages.

STEP 1 Make sure you do not have firewall running on your PC that could block the syslog port (port 514 by default).

STEP 2 Connect to the administration web server, and choose Admin access with Advanced settings.

STEP 3 To enable debugging, complete the following tasks:

a.Click Voice tab > System.

b.In the Debug Server field, enter the IP address of your syslog server. This address and port number must be reachable from the SPA9000.

c.From the Debug level drop-down list, choose 3.

STEP 4 To collect information about the type of NAT your router is using, complete the following tasks:

a.Click Voice tab > SIP.

b.Scroll down to the NAT Support Parameters section.

c.From the STUN Test Enable field, choose yes.

STEP 5 To enable SIP signalling, complete the following task:

a.Click Voice tab > Line N, where N represents the line interface number.

b.In the SIP Settings section, choose full from the SIP Debug Option field.

ATA Administration Guide

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Contents Administration Guide OL-17901-01 Contents About This Document Basic Administration and ConfigurationContents Configuring Your System for Itsp InteroperabilityConfiguring Voice Services Configuring Music on Hold Configuring the Pstn FXO Gateway on the SPA3102Appendix a ATA Routing Field Reference 111 Call Scenarios 105Appendix B ATA Voice Field Reference 121 Info 122Line 165 Pstn Line page SPA3102 190 Appendix F Where to Go From Here 244 Purpose AudiencePreface Product Firmware Version FirmwareDocument Conventions Typographic Meaning ElementChapter Contents OrganizationThese appendices provide information about other Press Enter Click Search ATA Administration Guide Introducing Cisco Small Business Analog Telephone Adapters Comparison of ATA Devices Introducing Cisco Small Business Analog Telephone AdaptersIntroducing Cisco Small Business Analog Telephone Adapters How ATAs Provide Voice Connectivity ATA Connectivity Requirements PAP2T Connectivity SPA2102 Connectivity SPA3102 Connectivity SPA8000 Connectivity SPA8000ATA Software Features Voice Supported CodecsAudio codec Other ATA Software Features SIP Proxy RedundancyLine tabs. See ATA Voice Field Reference, on page121 ATA Voice Field Reference, on page121 DtmfSee Configuring Dial Plans, on page 61 for more Tabs. See ATA Voice Field Reference, on page121 Voice Field Reference, on page121 Register Retry is configured in the SIP tab. See ATA Voice Introducing Cisco Small Business Analog Telephone Adapters Basic Administration and Configuration Basic Services and Equipment RequiredDownloading Firmware Basic Installation and Configuration Basic Administration and ConfigurationUpgrading the Firmware for the ATA Device Setting up Your ATA Device Using the Administration Web Server Connecting to the Administration Web Server Setting Up the WAN Configuration for Your ATA DeviceClick Network tab WAN Setup For Dhcp For Static IP AddressingFor PPPoE Registering to the Service Provider Upgrading, Rebooting, and Resyncing Your ATA Device Advanced ConfigurationsUpgrade URL Resync URL Reboot URL Provisioning Your ATA DeviceProvisioning Capabilities Configuration Profile Basic Administration and Configuration Configuring Your System for Itsp Interoperability Network Address Translation NAT and Voice over IP VoIPNAT Mapping with Session Border Controller Configuring NAT Mapping with a Static IP AddressConfiguring Your System for Itsp Interoperability NAT Mapping with SIP-ALG RouterClick Submit All Changes Configuring NAT Mapping with Stun Configuring Your System for Itsp Interoperability Click Voice tab System Configuring SIP Timer Values Firewalls and SIPSupported Codecs PAP2T / SPA2102 / SPA3102 / SPA8000Using a FAX Machine SPA2102, SPA3102 or SPA8000 Configuring Voice ServicesFAXPassthruMethod ReINVITE Fax Troubleshooting Parameter Tab Description and Value Managing Caller ID ServiceDefault is BellcoreN.Amer, China FSKCAS Silence Suppression and Comfort Noise Generation Configuring Dial Plans About Dial PlansDigit Sequence Function 2 3 4 5 6 7 8 9Digit Sequence Examples Local dialing with seven-digit number Blocked number Acceptance and Transmission the Dialed Digits Terminating Event ProcessingDial Plan Timer Off-Hook Timer Interdigit Long Timer Incomplete Entry Timer Syntax 1 Ss, dial plan Syntax 2 sequence SsEditing Dial Plans Click Voice tab RegionalConfiguring Voice Services Secure Call Implementation Enabling Secure CallsSecure Call Details Using a Mini-Certificate Generating a Mini Certificate Genmc ca-key user-name user-id expire-dateExample SIP Trunking and Hunt Groups on the SPA8000 About SIP Trunking Logical Block Diagram of SIP Trunking ItspSetting the Trunk Group Call Capacity Inbound Call Routing for a Trunk GroupContact List for a Trunk Group Syntax line,line,line…,hunt=hrule,cfwd=targetExamples 3,4,5,6,7,8,hunt=re*1Outgoing Call Routing for a Trunk Group ?,hunt=ra121,cfwd=14085550123Configuring a Trunk Group Trunk Group Management Setting the Hunt Policy Additional Notes About Trunk Groups Using the Internal Music Source for Music On Hold Using the Internal Music SourceConfiguring Music on Hold Changing the Music File for the Internal Music SourceConfiguring a Streaming Audio Server About the Streaming Audio ServerConfiguring Music on Hold Configuring the Streaming Audio Server Using the IVR with an SAS Line Configuring the Pstn FXO Gateway on the SPA3102 Connecting to Pstn and VoIP ServicesConfiguring the Pstn FXO Gateway on the SPA3102 How VoIP-To-PSTN Calls WorkOne-Stage Dialing Authentication Parameters Web Description Values Two-Stage Dialing How PSTN-To-VoIP Calls Work Parameters for Two-Stage Dialing Web Description ValuesTerminating Gateway Calls Parameter Web Description ValuesIs Medium VoIP Outbound Call Routing Configuring VoIP Failover to Pstn Example DescriptionSharing One VoIP Account Between the FXS and Pstn Lines Parameter Web Description ValueYes Other Options Pstn Call to Ring LineSymmetric RTP Call Scenarios Call Progress TonesVoIP to Pstn Call With and Without Authentication Pstn to VoIP Call with and Without Ring-ThruUsing Http Digest Authentication Without Authentication Call Forwarding to Pstn Gateway Forward-On-No-Answer to the Pstn GatewayForward to a Particular Pstn Number ATA Routing Field Reference Router StatusProduct Information section System Status sectionATA Routing Field Reference WAN Setup Internet Connection Settings sectionStatic IP Settings section PPPoE Settings sectionOptional Settings section DHCP, and DHCP/ManualOn or On when Phone is Use default MAC Clone Settings sectionQOS Settings section Remote Management sectionNetworking Service section LAN SetupVlan Settings section Vlan IDLAN Networking Settings section Static Dhcp Lease Settings sectionApplication Port Forwarding Settings section DMZ Settings sectionMiscellaneous Settings section System Reserved Ports Range sectionATA Voice Field Reference Info ATA Voice Field ReferenceLine Status section Idle ATA Administration Guide 125 System Information section PAP2T Pstn Line Status section SPA3102Dhcp Pstn Disconnect Tone VoIP State May take one of the following values Trunk Status section SPA8000 System Configuration section SystemOptional Network Configuration section PAP2T Internet Connection Type section PAP2TMiscellaneous Settings section not used with PAP2T Table, Manual/DHCP, and DHCP/ManualSIP SIP Parameters sectionDefault is application/dtmf-relay Default is application/hook-flashSIP T2 SIP Timer Values sec sectionSIP T1 SIP T4ATA Administration Guide 136 SIT2 RSC Response Status Code Handling sectionSIT1 RSC SIT3 RSCRTP Parameters section SIT4 RSCATA Administration Guide 139 SDP Payload Types section NAT Support Parameters section EXT IP ATA Administration Guide 143 Trunking Parameters section SPA8000 Regional Call Progress Tones section Default is 985@-16,1428@-16,1777@-1620.380/0 Default is 440@-19,480@-19*1/1/1+2Default is 600@-16 1.25/.25/1 Default is 914@-16,1371@-16,1777@-1620.274/0Distinctive Ring Patterns section Distinctive Call Waiting Tone Patterns section Default is 30.3/.1,.3/.1,.1/9.1 Default is 30.1/.1, .3/.1, .1/9.3Distinctive Ring/CWT Pattern Names section Control Timer Values sec section Ring and Call Waiting Tone Spec sectionATA Administration Guide 152 Vertical Service Activation Codes section ATA Administration Guide 154 ATA Administration Guide 155 ATA Administration Guide 156 ATA Administration Guide 157 ATA Administration Guide 158 Vertical Service Announcement Codes section SPA2102, SPA8000 Outbound Call Codec Selection Codes sectionATA Administration Guide 160 Miscellaneous section 600, 900, 600+2.16uF, 900+2.16uF, 270+750150nF220+850120nF, 220+820115nF, or 200+600100nF ATA Administration Guide 162 ATA Administration Guide 163 Caller ID Method Following choices are available Use, bell 202 orLine Line Enable section Streaming Audio Server SAS sectionNAT Settings section Network Settings section Or disableMedium, high, very high, or extremely high SIP Settings section Field DescriptionSIP Guid Configured Proxy or Outbound Proxy if Use Outbound Call Feature Settings section FromURL Proxy and Registration section Outbound Proxy and Use OB Proxy in Dialog parametersSubscriber Information section Supplementary Service Subscription section ATA Administration Guide 176 ATA Administration Guide 177 Audio Configuration section Gateway Accounts section SPA3102VoIP Fallback to Pstn section SPA3102 Dial Plan sectionDial Plan Entry Functionality Default is *xx3469110002-9xxxxxx1xxx2 9xxxxxxS0xxxxxxxxxxxxFXS Port Polarity Configuration section Trunk Group page SPA8000Voice tab Trunk Group ATA Administration Guide 183 ATA Administration Guide 184 ATA Administration Guide 185 Inbound calls When the limit is reached, the Trunk SUA Number cfwd=14089993326 #0-9A-D ATA Administration Guide 189 Pstn Line page SPA3102 Voice tab Pstn Line ATA Administration Guide 192 ATA Administration Guide 193 This field is not available with the PAP2T. The Global Proxy SIP proxy server for all outbound requests Outbound Proxy parameter and Use OB Proxy in Dialog is ATA Administration Guide 197 G726-16,G726-24,G726-32,G726-40,G729a, or G723 Info ATA Administration Guide 200 Dial Plans section VoIP-To-PSTN Gateway Setup section Caller 1/2/3/4/5/6/7/8 PIN VoIP Users and Passwords Http Authentication section Ring Settings section FXO Pstn Timer Values sec sectionATA Administration Guide 206 Pstn Disconnect Detection section Detect Long Silence is yes Impedance ATA Administration Guide 210 International Control Settings section ATA Administration Guide 212 User Call Forward Settings section Selective Call Forward Settings section Speed Dial Settings sectionSupplementary Service Settings section Distinctive Ring Settings section Default is New VM Available PSTN-To-VoIP Selective Call Forward Settings section PSTN-To-VoIP Speed Dial Settings sectionPstn User page SPA3102 Only Pstn Ring Thru Line 1 Distinctive Ring Settings section Pstn Ring Thru Line 1 Ring Settings sectionWlwmenoack No Feature/XML Tag Parameters ExamplesWlqos RtqosRtsp IgmpUpnp Qospriorityrule QOS PriorityruleBASICSET1 WLBASICSET1Wlbasicset WLBASICSET2Wlsecurity WLSECURITYSET1 WlsecurityWL SECURITYSET1 LAN Dhcp LandhcpWantype Fail Pattern Internet Connection Type WAN DNS Singleport Forwarding You also should configure a Dhcp Routersyslo G Troubleshooting How do I access the ATA device if I forget my password? TroubleshootingHow do I save my current configuration? How do I debug my ATA device? Is there a syslog?ATA Administration Guide 237 ATA Administration Guide 238 Environmental Specifications Environmental Specifications SPA2102SPA3102 SPA8000 WRP400 WRTP54GWRTP54G Product Resources Resource LinkWhere to Go From Here Related DocumentationDocument Title Description Intended Audience Document Title Description Intended Audience Additional Information Resource LocationSupport Contacts
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PAP2T, SPA8000, SPA3102, WRP400, SPA2102 specifications

The Cisco Systems SPA2102 is a versatile Voice over Internet Protocol (VoIP) adapter that serves as a bridge between traditional telephony systems and modern IP networks. Designed primarily for small to medium businesses, the SPA2102 is highly regarded for its reliability, ease of use, and rich feature set. This device allows users to make and receive phone calls over the internet while maintaining the ability to connect traditional analog phones.

One of the standout features of the SPA2102 is its dual-port capability. The device includes two FXS ports, allowing users to connect two separate analog telephones. This makes it an ideal choice for businesses that want to retain their existing telephony infrastructure while transitioning to a VoIP system. The ability to utilize two telephone lines simultaneously provides flexibility and convenience, especially for users in a busy office environment.

The SPA2102 leverages Session Initiation Protocol (SIP) technology, which is widely recognized for its robustness and interoperability. The support for SIP ensures that the SPA2102 can work seamlessly with various VoIP service providers, offering users a broad range of options for their telecommunication needs. In addition to SIP, the device supports various codecs, including G.711, G.726, and G.729, allowing for flexible audio quality settings and bandwidth management.

Security is also a critical aspect of the SPA2102. It incorporates advanced encryption methods, such as Secure Real-time Transport Protocol (SRTP) and Transport Layer Security (TLS), to protect voice communications from potential eavesdropping. This focus on security makes the SPA2102 a reliable choice for businesses concerned about the confidentiality of their conversations.

The device is easy to configure and manage, thanks to its web-based interface. This allows administrators to quickly set up the adapter, manage network settings, and troubleshoot any issues that may arise. Furthermore, the SPA2102 supports Quality of Service (QoS) features, ensuring that voice traffic is prioritized over other types of network traffic, which enhances call quality and reliability.

Overall, the Cisco SPA2102 is a powerful, user-friendly VoIP adapter that combines traditional telephony with modern IP technology. Its dual-port capability, support for SIP, extensive security features, and ease of configuration make it a solid choice for businesses looking to innovate their communication systems while minimizing disruption. Whether used in a small office or a larger organizational setting, the SPA2102 continues to be a reliable component of VoIP solutions.
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