LogoLogo
Help
  • Overview
    • Introduction
    • Characteristics
    • Platforms
  • INITIAL Installation
    • 1. Instructions by platform
      • ProSBC Requirements Matrix
      • Baremetal Installation
        • List of Supported Network Interface Cards
        • SBC Certified Hardware
          • High Performance Baremetal Server
          • Medium Performance Baremetal Server
          • Ciena 3906mvi server for Customer Premises Equipment (CPE)
          • Qotom Barebone server for Customer Premises Equipment (CPE)
          • Lanner Barebone server for Customer Premises Equipment (CPE)
          • Telco Systems virtualization platform on Lanner NCA-2510 server for Customer Premises Equipment(CPE)
      • Virtual, self-hosted Installation
        • Virtual - Proxmox
        • Virtual - VMware
          • Launching an Instance of VMware vSphere
          • Deploying ProSBC on VMware
          • Adding Network Interfaces in VMware
          • Configuring Passtrough interfaces on VMware
      • Cloud Installation
        • Cloud - AWS
          • AWS Installation
          • Cloud Formation Installation
          • Instance Upgrade
          • AWS Additional Interface
          • AWS Installation Troubleshooting
          • Recovering an Elastic IP address
        • Cloud - Azure
    • 2. Initial Configuration
      • Initial Setup
        • SBC Management IP configuration
      • Basic configuration
        • Configuring IP interfaces
        • Creating a SIP stack
        • Creating a SIP transport server
        • Allocating an SIP NAP
        • Allocating a SIP open NAP
        • SIP Transport DNS settings
        • Creating a first call route
    • 3. Uploading a License
      • Host-control
        • SELinux
        • SELinux management
      • Add/Change Licenses
        • Add/Change Licenses Manually
  • Use Cases
    • Applications
      • Carrier Interconnection
      • Monitoring as a Service (MaaS)
      • NGN Interconnection
      • Operator Interconnection
      • SIP subscribe notify publish forwarding
      • STIR/SHAKEN
      • Transcoding
      • SIP Trunking
      • Hosted PBX
      • SIP Network Peering
      • Remote Workers
    • Interoperability Examples
      • STIR/SHAKEN with Transnexus and ClearIP
      • Fraud Detection [YouMail]
      • Skype Connect
      • Skype for Business S4B TCP
      • Skype for Business S4B TLS
      • Asterisk
      • 3CX
      • FreePBX
      • FusionPBX
      • FreeSWITCH
      • Twilio
      • Sippy
      • Avaya IP Office
      • Cisco UCM 12
      • Brekeke PBX
      • VitalPBX
      • Yeastar P-Series Cloud
      • VoIP.ms
      • Wildix
  • CONFIGURATION DETAILS
    • Configuration By Web Portal Category
      • System Settings
        • Setting the Role to Standalone
        • Setting the Role to a Primary Unit in a 1+1 System
        • Setting the Role to a Secondary Unit in a 1+1 System
        • Resetting the Host Role
        • Resetting the Network Device Role
        • Create Session Border Gateway Access Control List (ACL)
        • Session Border Gateway: Advanced Parameter Settngs
        • Create Session Border Gateway Access Control List (ACL) Filters
        • Connecting to the Web Server and Logging on to the Web Portal
        • Logging Off
        • Modifying Security Settings
        • Creating Web User groups
        • Creating Web Users
        • Modifying Web User Permissions
        • Enabling and Disabling a User
        • Deleting a User
        • Accessing Audit Logs
        • Activating the Configuration
        • Configuring a Web Portal Profile
        • Configuring the Date, Time, Timezone and NTP servers
        • Configuring the DNS
        • Create HTTP Service
        • Use HTTPS service
        • Configure HTTPS certificates
        • Configuring letsencrypt certificate
        • Configuring the ICMP
        • Configuring the SSH
        • Upgrade Telcobridges linux software packages
        • Retrieving a Software Release
        • Uploading a Software Release
        • Activating a Software Release
        • Retrieving a License
        • Uploading a License
        • Database Backup
        • Downloading a Database Backup
        • Uploading a Database Backup
        • Restoring a Database
        • Enabling the SNMP Agent
        • Configuring the SNMP Agent
        • Creating an SNMPv1/SNMPv2 Community
        • Creating an SNMPv3 User
        • Creating an SNMP Trap Destination
      • IP Network Settings
        • Configuring a Virtual Port
        • Configuring a VLAN
        • Configuring an IP Port Range
        • Configuring IP Interfaces
        • Configuring NAT Traversal
          • Local NAT Traversal
          • Remote NAT Traversal
        • DNS Configuration
          • Creating a DNS Local Entry
        • Configuring VoIP Interfaces
      • SIP
        • Creating a SIP Stack
        • Creating a SIP Transport Server
        • TLS/SRTP
          • Creating TLS Certificates
          • Adding TLS Certificates
          • Configuring TLS Profiles
        • Enabling SIP-I/SIP-T
        • SIPREC Forwarding
      • SIP Registrar
        • Creating a SIP Domain
        • Creating a SIP Registrar
        • Creating a SIP Register Filtering Rule
        • Creating a SIP Register Filtering Rule Condition
        • Creating a SIP Register Filtering Rule Action
      • Network Access Points (NAP)
        • Allocating a SIP Open Network Access Point (NAP)
        • SIP NAP Polling
      • NAP Profiles
        • Profile SDP Description
        • Fax Settings
          • Configuring Fax Relay
          • Configure Fax Passthrough
          • Configure Fax T38
          • Configure Fax NSE
          • Configure Fax VBD
      • Call Routing
        • Creating a First Call Route
        • Enable Flexible NOA Routing Script
        • Add NOA Columns in Routes
        • Import Customized Routing Script
        • Add Customer Column in Routes
        • Add Customized Filter Script To Main Script
        • Adding Label Routing to a Routing Script
        • Assign Routing Script Database Files to the Gateway Application
        • Add Digitmap Files to the System
        • Add Routeset Definition Files to the System
        • Assign Definition Digitmap Files on a per NAP Basis
        • Generate Dynamic Routes
        • Steps to configure label routing for Group of DIDs to a single outbound NAPs
        • Steps to configure label routing for Group of DIDs to multiple outbound NAPs
        • Group of DIDs to multiple outbound NAPs: Load-sharing mode
        • Group of DIDs to multiple outbound NAPs: Priority Mode
        • Update the Digitmap Files
        • Update the Routeset Definition Files
        • Configuring RADIUS Authorization
        • Importing a RADIUS Custom Dictionary
      • Lawful Intercept
        • Lawful Intercept Status
        • Verifying lawful interception
        • Importing a Lawful Interception .CSV File
        • Enabling Lawful Interception in a Routing Script
        • Configuring Lawful Interception
      • Call Detail Records (CDR)
        • CDR Variables
          • Call statistics format
        • Retrieve Text CDRs
          • Automatic CDR Retrieval
        • RADIUS CDRs
          • Configuring RADIUS
          • Adding RADIUS Server(s)
          • RADIUS CDR attributes
      • Routing Scripts
        • Development Guides & Tutorials
          • Accessing Routing Script Parameters
          • Parameter Mapping
          • Script Parameters Definition
          • Script Parameters Definition for SIP
          • Accessing Information about Registered Users
          • Route Parameters and Call Routing
          • Playing prompts, announcements, and tones
          • Recording
          • User-to-User Information
          • Radius Authorization
          • ENUM Query
          • DNS Query
          • Call Diversion Options
          • Call Transfer Requests
          • Redirection
          • Connect Number
          • Terminating Calls
          • NAP Status and other NAP Information
          • Telephony Services (CNAM Requests over SS7)
          • Custom User Context
          • Routing Script Tests
          • Create New Routing Script
          • Enable Routing Script
    • Configuration By Use Case
      • SIP Trunking Configuration
        • Configuration Files for SIP Trunking Scenario
        • SIP Trunk Configuration Instruction with 3CX
        • SIP Trunk Configuration Instruction with FreePBX
        • SIP Trunk Configuration Instruction with FusionPBX
        • SIP Trunk Configuration Instruction with FreeSWITCH
        • SIP Trunk Configuration Instruction with Twilio Elastic trunking
        • SIP Trunk Configuration Instruction with Avaya IP Office
        • SIP Trunk Configuration Instruction with Brekeke PBX
        • SIP Trunk Configuration Instruction with Avaya IP Office
        • SIP Trunk Configuration Instruction with Yeastar P-Series Cloud
        • SIP Trunk Configuration Instruction with Cisco UCM
        • SIP Trunk Configuration Instruction with VoIP.ms SIP trunking
        • SIP Trunk Configuration Instruction with Wildix Cloud VoIP PBX
        • Configuration for Adding ProSBC as a SIP Trunk in the FreePBX Server
        • FreePBX Extension Creation
        • FusionPBX SIP Trunk Creation
        • FusionPBX Extension Creation
        • FreeSWITCH SIP Trunk Creation
        • Twilio Elastic SIP Trunking Configuration
        • Sippy SIP Trunk Creation
        • Avaya IP Office Trunk Creation
        • Cisco UCM 12 Trunk Creation
        • Adding ProSBC as a SIP Trunk in the Brekeke PBX
        • VitalPBX Extension Creation
        • Adding ProSBC as a SIP Trunk in the Yeastar P-Series Cloud
        • Adding ProSBC as a SIP Trunk in the Wildix Cloud VoIP PBX
        • SIP Trunk Configuration Instruction with VitalPBX
        • VitalPBX SIP Trunk Creation
      • Configuring SIP Registration to SIP Proxy
      • Configuring a Hosted PBX
      • Multiple Domains/Hosted PBXs
      • SIP Network Peering / IP Carrier Interconnection
      • Remote Workers
        • Configuration Files for Remote Office/Workers
        • Remote Workers Configuration Instruction with FusionPBX
        • Remote Workers Configuration Instruction with 3CX
        • Remote Workers Configuration Instruction with FreePBX
        • Remote Workers Configuration Instruction with VitalPBX
      • ProSBC and ClearIP (TransNexus)
        • Configuration for STIR/SHAKEN with Transnexus' ClearIP service
        • Configuration for CNAM Verification and Robocall Analytics with Transnexus' ClearIP service
        • Configuration for Robocall Mitigation with Transnexus' ClearIP service
        • Configuration for 302 Redirect routing with Transnexus' ClearIP service
        • Configuration for CAPTCHA Authentication – 302 Redirect with Transnexus' ClearIP service
        • Configuration for STIR/SHAKEN with Transnexus' ClearIP service
      • Transcoding Unit Configuration
        • Baremetal and Virtual Machine
        • Show the Hardware Units menu
        • Adding Transcoding Unit
      • Configuration for Adding YouMail Script to Routing Scripts
      • Skype Connect Example Configuration
      • Skype for Business Example Configuration
      • 3CX Phone Provisioning Configuration
        • Configuration for 3CX PBX Server with the ProSBC to receive T38 Faxes
        • Configuration for 3CX PBX Server with the ProSBC as SIP trunk
      • SIP Emergency
      • SIP registration forwarding
        • Creating a SIP Domain
        • Configuring SIP Registration for Open NAP
        • Configuring SIP Registration for regular NAP
      • RTP no-anchoring
        • Parameter: Allow low-delay media relay
          • Configuring an IP Port Range
        • Creating Profiles
          • Modifying SDP Profile Settings
          • Modifying SIP Profile Settings
          • Modifying RTP and Audio Settings
          • Modifying FAX Relay Profile Settings
          • Modifying Telephony Profile Settings
          • Modifying Tones and Call Progress Options
          • Modifying IVR Record Profile Settings
          • Modifying LNP Profile Settings
          • Modifying Multilevel Precedence and Preemption (MLPP) Options
          • Modifying Call Transfer Profile Settings
          • Modifying Tone Definition Profile Settings
    • Configuration Parameters (all)
    • Routing Script - SIP 302 Handling
  • Maintenance & Troubleshooting
    • Maintenance Guide
      • Check Disk Space
      • ProSBC Processor Usage
      • Troubleshooting Toolpack
      • Restoring a Database
    • System Upgrades
      • Migrate current database
      • Upgrade Telcobridges linux software packages
    • Software version release notes
    • Software version release download
    • ProSBC public roadmap
  • Troubleshooting & Support
    • Troubleshooting Tips & Actions
      • Configuring Call Trace
        • Retrieving Call Trace
        • Call Trace Filter Parameters
      • Creating a test call
      • tbsigtrace: Signaling trace capture tool
        • Accessing Device
          • TMG:Change Management IP Address
          • Password less ssh
          • How to setup ssh tunnel with PuTTY
        • Live Signaling Capture with tbsigtrace
      • How to Get Rid of Sub Optimal Warning
      • How to Lower The Trace Level on an Application
      • TBReport
      • VoIP Ethernet Capture on a ProSBC
      • Enabling Call Recording
      • Accessing the Call Recording
      • Routing Scripts
        • Update Your Routing Scripts
        • Disabling a Call Route
    • Troubleshooting Common Problems
    • Support Links
      • Support Forums
      • ProSBC Training
      • Customer Dashboard User Guide
      • Contacting TelcoBridges technical support
      • Frequently Asked Questions
      • Sending Large Files to TelcoBridges
    • How to use tbx cli tools remote program
  • Tools, Tips, and Tricks
    • TelcoBridges Magic Bookmark
    • Video Library
    • RESTful API
      • Postman Examples
      • Ruby Examples
      • TBConfig Examples
        • Exporting a Configuration
        • Importing a Configuration
        • Activating a Configuration
        • Updating a Route
        • Dropping Calls
      • ProSBC:Restful API SIP Domain
      • ProSBC:Restful API SIP Domain Registrar
      • Extracting Call Traces with the API
    • TBStatus API
      • Tbstatus monitoring
      • Status API
      • Dropping calls
  • Appendices
    • Appendix A: Glossary
      • Glossary: Call Detail Records (CDR)
      • Glossary: Call routing
      • Glossary: DNS
      • Glossary: Mean Opinion Score (MOS)
      • Glossary: NAP
      • Glossary: RADIUS
      • Glossary: Ringback tones
      • Glossary: SAP
      • Glossary: Signaling protocols
      • Glossary: SIP
        • Glossary: Route retry
        • Glossary: SIGTRAN
        • Glossary: SIP-I/SIP-T
        • Glossary: SIP gateway
        • Glossary: SIP Registration
      • Glossary: Softswitch
      • Glossary: Toolpack
        • Glossary: Web server
        • Glossary: tboamapp
          • Glossary: Tbtoolpack Service
            • System Id
              • Gateway Port
          • Primary/Secondary
          • Master/Slave
            • Active/Standby
      • Glossary: Unified communications
      • Glossary: Web Portal
      • Glossary: DTMF Relay
    • Appendix B: Product Datasheets
Powered by GitBook
On this page

Was this helpful?

  1. CONFIGURATION DETAILS
  2. Configuration By Web Portal Category
  3. NAP Profiles

Profile SDP Description

PreviousNAP ProfilesNextFax Settings

Last updated 8 months ago

Was this helpful?

You can specify the Profile SDP Description to define which are supported in the using this Profile.

Here is the default Profile SDP Description.

m=audio 0 RTP/AVP 0 8 4 18 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,32-36

Each line of the Profile SDP Description consists of text of the form <type>=<value>. <type> is always exactly one character and is case-significant. <value> is a structured text string whose format depends on <type>. It also will be case-significant unless a specific field defines otherwise. Whitespace is not permitted either side of the `=' sign. In general <value> is either a number of fields delimited by a single space character or a free format string.

In our Profile SDP Description, only 2 <type> are used.

Media Announcement

The format of media announcement is as follows.

m=<media> <port> <transport> <fmt list>

  • The first sub-field is the media type. Currently defined media for TMedia is "audio".

  • The second sub-field is the transport port to which the media stream will be sent. In Toolpack, it is not specified in Profile SDP Description and therefore you should specify "0".

  • The third sub-field is the transport protocol. For most of the application, you may specify it as "RTP/AVP" - the IETF's Realtime Transport Protocol using the Audio/Video profile carried over UDP.

  • The fourth and subsequent sub-fields are media formats. For audio and video, these will normally be media payload types as defined in the RTP Audio/Video Profile. When a list of payload formats is given, this implies that all of these formats may be used in the session, but the first of these formats is the default format for the session. When the transport protocol is specified as "RTP/AVP", the payload format can be specified as either

    • the payload type number for static payload types

    • the payload type number along with additional encoding information for dynamically allocated payload types.

The payload type, which is carried in the actual RTP packet header, is used to identify the type of codec information carried in the packet. A list of payload type values for each codec is defined within RFC3551. Unfortunately, since the payload type field is only 7 bits-wide, all codecs cannot have a permanent payload type value understood universally by all VoIp systems. Therefore, some codecs have dynamic values that need to be negotiated through a call control or session control protocol such as SIP before the actual RTP session can take place.

Here is the list of codec payload type values per RFC3551.

Codec
Payload type value

G.711 uLaw

0

G.723.1

4

G.711 aLaw

8

G.722

9

Comfort Noise

13

G.728

15

G.729AB

18

G.726-40

dynamic

G.726-32

2 or dynamic (depends on the network)

G.726-24

dynamic

G.726-16

dynamic

G.729EG

dynamic

AMR

dynamic

EVRC

dynamic

QCELP

dynamic

When you use dynamic payload types, you need to specify the additional encoding information using the attribute for media announcement.

Attribute for Media Announcement

A media description may have any number of attributes ("a=" fields) which are media specific. The format of attribute is as follows.

a=ora=:<value>

Here are some examples of attributes.

Dynamic payload type

You specify the additional encoding information for dynamic payload type in the following format:

a=rtpmap:<payload type> <encoding name>/<clock rate>[/<encoding parameters>]

For audio streams, <encoding parameters> may specify the number of audio channels. This parameter may be omitted if the number of channels is one provided no additional parameters are needed.

Other media specific attribute

The use of other media specific attributes depends on the specification of the RTP payload format for the specific media type. Here are some example of attributes.

  • a=fmtp:<format> <format specific parameters>

This attribute allows parameters that are specific to a particular format to be conveyed in a way that SDP doesn't have to understand them. The format must be one of the formats specified for the media. Format-specific parameters may be any set of parameters required to be conveyed by SDP and given unchanged to the media tool that will use this format.

  • a=ptime:<packet time>

This gives the length of time in milliseconds represented by the media in a packet. This is probably only meaningful for audio data. It should not be necessary to know ptime to decode RTP or vat audio, and it is intended as a recommendation for the encoding/packetisation of audio. It is a media attribute, and is not dependent on charset.

Examples

Here are some examples of the attributes found in the default profile SDP description.

a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20

The above defines iLBC codec of 8000/sec sample rate with 20ms frame size.

 a=rtpmap:98 telephone-event/8000

The above defines the DTMF and telephony tones relay over RTP using RFC2833.

Here is another example, enabling Voice Activity Detection (VAD) for G.711 ulaw, G.711 alaw, G.723.1a and G.729b. It also enable relay of DTMF events 0 to 15 and telephony tones events 32 to 36.

m=audio 0 RTP/AVP 0 8 4 18 98 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6300;annexa=yes
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-15,32-36
a=rtpmap:13 CN/8000

Another example to disable silence suppression explicitly in the SDP.

m=audio 0 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,32-36
a=silenceSupp:off - - - -

Below shows an example to enable GSM-FR with a dynamic payload type 99:

m=audio 0 RTP/AVP  18 4 8 0 96 97 98 99 13
a=rtpmap:96 iLBC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:98 telephone-event/8000
a=rtpmap:99 GSM-FR/8000
a=fmtp:4 bitrate=6.3
a=fmtp:4 annexa=no

Note: 1. Your toolpack license needs to be GSM enabled. Please contact TelcoBridges support if you are not sure. 2. You may need additional third party GSM license(s) from the corresponding patent holders. e.g. GSM-EFR, GSM-HR

Below shows an example to enable GSM-FR with a dynamic payload type 99:

m=audio 0 RTP/AVP  18 4 8 0 96 97 98 99 13
a=rtpmap:96 iLBC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:98 telephone-event/8000
a=rtpmap:99 GSM-FR/8000
a=fmtp:4 bitrate=6.3
a=fmtp:4 annexa=no

This example is G.711 alaw with a 10ms packet (instead of the default 20ms) :

m=audio 0 RTP/AVP  8
a=rtpmap:8 PCMA/8000
a=ptime:10

This example is AMR-WB (using dynamic payload type 113) and G.722 (static payload type 9) :

m=audio 0 RTP/AVP  113 9 
a=rtpmap:113 AMR-WB/16000

References

- SDP: Session Description Protocol

- RTP Profile for Audio and Video Conferences with Minimal Control

- Real-time Transport Protocol (RTP) Payload Format for internet Low Bit Rate Codec (iLBC) Speech

- RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

voice codecs
NAP
RFC2327
RFC3551
RFC3952
RFC2833