System Application Services

All the sipXecs application services are allocated to specific server roles. Using the centralized cluster management system each role can be instantiated on a dedicated server or several (all) roles can be run on a single server. Configuration of all services and participating servers is fully automatic and Web UI based.

SIP Session Router, optionally geo-redundant and load sharing
Media server for unified messaging and IVR (auto-attendant) services
Conferencing server based on FreeSWITCH
XMPP Instant Messaging (IM) and presence server (based on Openfire)
Contact center (ACD) server
Call park / Music on Hold (MoH) server
Presence server (Broadsoft and IETF compliant resource list server for BLF)
New: Shared Appearance Agent server to support shared lines (BLA)
Group paging server
SIP trunking server (media anchoring and B2BUA for SIP trunking & remote worker support)
Call Detail Record (CDR) collection & processing server
Third party call control (3PCC) server using REST interfaces
Management and configuration server
Process management server for centralized cluster management

SOA Architecture / Business Process Integration using Web Services

Web Services SOAP interfacefor key administrative functions
Web Services REST interface for user portal functions and third party call control
All components centrally managed using XML RPC
Google Web Toolkit (GWT)

Core Calling Features (Telephony Features)

Transfer (consultative & blind)
Call coverage
Call hold / retrieve
Consultation hold
Music on Hold for IETF standards compliant phones
User-specific MoH files
MoH music from an external streaming source
Admin or user configurable Busy Lamp Field (BLF) presence and softkeys
New: Shared lines (BLA) for boss - secretary applications (Polycom only)
Uploadable music file
3-way / 5-way video and voice conference on the phone
Call pickup (global and directed call pickup)
Call park & retrieve
Hunt groups
Intercom with auto-answer (bi-directional)
SIP URI dialing
CLID (Calling Line Identification)
CNIP (Calling party Name Identification Presentation)
CLIP (Call Line Identification Presentation)
CLIR (Call Line Identification Restriction)
Per gateway CLIP manipulation
Call waiting / retrieve
Do not Disturb (DnD)
Forward on busy, no answer, do not disturb
Multiple line appearances
Multiple calls per line
Multiple station appearance
Outbound call blocking - Calls from phones to PSTN numbers, or classes of numbers, can be blocked based on:
- The destination of the call; for example, when a user or device cannot initiate an international long distance call.
- The source of the call; for example, when a lobby phone can only initiate calls to internal numbers.
Click-to-call
Redial
Call history (dialed, received, missed)
Auto off-hook / ring down
Incoming only
Configuration of individual Speed Dial softkeys
Auto-generation of directory information

E911 Emergency Response

Internal notification using email and SMS

Remote Branch office support

Centralized deployment: Branch only provides phones and optionally PSTN gateway for failover, reduced WAN BW consumption or E911 calls
Distributed deployment: Branch provides full call server with SIP site-to-site dialing between offices
Branch office locations can be defined in the mgmt UI with a postal address
Users, phones, gateways, SBC, and servers can be assigned to a branch location
A PSTN gateway can be available for calls that originate in a specific branch only or for general use
Source routing allows call routing based on location (branch local calls are routed through local gateway preferably)
Branch postal address automatically proliferates to user's office address
Survivable branch configuration possible with Audiocodes gateways SAS functionality (auto-configured)
Certain sipXecs services can be deployed in the branch as part of the cluster (e.g. conferencing)
(for a future release we will add full survivable branch capability based on the sipXecs cluster - see XX-4819)

Enterprise Instant Messaging (IM) and Presence

XMPP based IM and presence server based on Openfire
New role in sipXecs cluster management for IM/presence
Supports XMPP standards based clients
Auto-configuration of user's IM accounts
Auto-configuration of IM user groups
Personal group chat room for every user auto-configured
Federation of phone presence with IM presence
Customizable "on the phone" presence status message
dynamic call routing based on user's presence status
Message archiving and search for compliance (pending)
Server-to-server XMPP federation
Optional secure client connections
Client-to-client file transfer
Group chat rooms
XMPP search
Integration of user profile information and avatar (pending)

Personal Assistant IM Bot

Personal Assistant feature
Dynamic call control using IM
Dynamic conference management using IM
Unified messaging management using IM
Enables Fixed Mobile Convergence (FMC) application
Call history / missed calls
Call initiation using corporate dialplan
Corporate directory look-ups

Presence and IM Federation

Server side federation with other public IM systems (based on Kraken IM Gateway) including:
Yahoo, AIM, MSN
- ICQ, IRC
- IBM Sametime
- Facebook IM
- MySpace IM
SIP / SIMPLE
Server-to-server federation with Google Talk
Allows group chat sessions across systems
Allows message archiving (if enabled) across systems
User self-administration of credentials for other IM systems

Fixed Mobile Convergence (FMC) Application

FMC application with the following functionality:
- Enterprise number dialing
- System call-back saves on wireless toll charges
- Corporate directory look-ups
- Call history
- Presence sharing
- IM
Server-to-server (XMPP) federation with GTalk allows using GTalk client on smart phones with sipXecs
Dynamic conference bridge control

Web Conferencing & Collaboration

Since Salesforce.com acquisition Dimdim is no longer an open product. Therefore starting with sipXecs 4.4 version Dimdim integration was removed.

Integration with DimDim for Web conferencing
Automatically launches DimDim with correct voice conferencing info
Creates link to conference that can be sent to participants
Allows easy setup of DimDim personal account / stores DimDim credentials

User Self-Control (User Web Configuration Portal)

Every user on the system gets access to a personal Web user portal for self-management and control
Management of unified messaging (voicemail)
Configuration of unified messaging preferences
Time based find-me / follow-me
Flexible configuration of call forwarding
Management of personal profile data including avatar
Personal call history
Personal phone book, speed dial and presence management
Allows contact upload from GMail and Outlook
Click-to-call
ACD presence and supervision capabilities
Individual phone management
Personal auto-attendant
Dynamic conference management w/ click-to-conference
Management of personal IM account
Setup of DimDim Web conferencing
Personal MoH music upload and preferences

Superior Voice Quality

Peer-to-peer media routing for best quality (media not routed through the sipXecs server)
Unmatched voice quality with lowest delay and jitter
Support for any codec supported by the phone or gateway (including video)
Support for HD Voice (Polycom and other phones)
Codec negotiation (no transcoding required)
Conferencing, auto-attendant and voicemail support HD voice w/ transcoding if necessary

User Management

Create a user, provision a phone and assign a line in only three clicks - easy!
Numeric or alpha-numeric User ID
User PIN management (UI or TUI)
Aliasing facility (numeric and alpha-numeric aliases)
Extension and alias uniqueness assurance
Management or auto-assignment of user's IM ID and display name
Automatic IM buddy list creation based on user groups
Granular per user permissions
Call permissions:
- 900 Dialing
- International Dialing
- Long Distance Dialing
- Mobile Dialing
- Local Dialing
- Toll Free Dialing
- Forward Calls External
System permissions:
- User has voicemail inbox
- User listed in auto-attendant directory
- User can record system prompts
- User has superuser access
- User allowed to change PIN from TUI
- User can use Microsoft Exchange 2007 VM
- User has a personal auto-attendant
Custom permissions as defined by the admin
Supervisor permission for groups (e.g. Call Center supervisor)
Management of user contact record (user profile)
Comprehensive profile data
Work and home address
In-building location information
Assistant information
Support for avatar including support for gravatar
SIP password management for security
User groups with group properties
Per user call forwarding (follow me)
- To local extension, PSTN number, or SIP address
- Based on user or admin defined time schedules
- Parallel or serial ring
- Allows definition of ring time before trying next number
- Allows several forwarding destinations
- Follow-me configuration using user portal
Extension pool with automatic assignment
Per user Caller ID (CLID) assignment
Per user Caller ID blocking

Dial Plan

Easy to use GUI based dial plan manipulation
Time-based dialing rules with different admin defined schedules
Rules based least cost routing
Dynamic call routing based on user's IM presence status
Directly route to voicemail on IM status DND
Dynamically add forwarding destination based on phone number in custom presence status
Automatic gateway redundancy and fail-over
Specific E911 routing
Permission based rules
Prefix manipulation
Dialplan templating for international dial plans
Built-in support for U.S., German, Swiss, and Polish local dial plans
(Any other local dial plan can be added as a plugin)
Specify internal extension length
ISN dialing based in ITAD numbers. See freenum.org
ENUM support
Specific rule for site-to-site call routing between SIP systems
Redirector plugins - any imaginable dial rule can be added as a plugin

Internet Calling

Ability to configure SIP URI based call routing to other domains
Specific SBC selection for call routing
Configuration of native NAT traversal w/ optionally redundant media anchoring if necessary
Media anchoring supports voice and video for any codec

Directory, Softkeys, Speed Dial

Automated generation of directory information per user or per user group
Support for complete contact information and user profile, including avatar
Crreation and Management of many different directories (per user, per user group, per location, etc.)
Upload of contacts from GMail and Outlook
User management of directory information
Automated provisioning of directory information into user's phones
Allows adding contacts to the directory from a .csv file (Excel)
User configurable speed dial (internal / external numbers, SIP URIs)
Speed dial generated server side and backed up
Auto-provisioning of speed dial to phones
User configuration of Busy Lamp Field (BLF) to monitor presence of other users or phones (e.g. attendant console)

PSTN Trunking

Unlimited number of PSTN gateways and trunk lines
Supports any SIP compliant gateway (e.g. Cisco, Audiocodes, Mediatrix, Vegastream, Patton, etc.)
Gateways can be in any location
Gateway selection per dialing rule
Source routing of calls so that calls can be routed through a local gateway to save WAN bandwidth
DID
Local DID per gateway
DNIS
CLIP Management
- User CLIP
- Gateway default CLIP
- Prefix stripping / appending
Per gateway CLIR
Automatic Route Selection (ARS)
- Implemented with XML-formatted mapping rules.
- Mapping values re-write SIP URLs to specify the next hop or destination for a SIP message that has been received by the Communications Server component.
- Direct messages to different SIP/PSTN trunk gateways, either on premise or at a remote premise location, based on any portion of SIP URL or E.164 number.
- Route messages to commercial SIP/PSTN service providers, which reduces or eliminates the need for on-premise trunk gateways.
Least-cost routing (LCR)
Automatic failover if unavailable
Automatic failover if busy
FAX support
Mixing of PSTN and SIP trunks with least cost routing

SIP Trunking

SIP trunking gateway w/ NAT traversal
Remote worker support w/ near-end and far-end NAT traversal and auto-detection
ITSP templates for simplified configuration
Interop with the following ITSPs:
BT (UK)
AT&T
Skype
Bandwidth.com
CBeyond
Bandtel
CallWithUs
Eutelia (Italy)
LES.NET
SIPcall (Switzerland)
Vitality
VOIPUser (UK)
VOIP.MS
Voxitas

Easy configuration templates exist for the above ITSPs
Many other ITSPs are compatible, see ITSP interop

SIP interop with Nortel CS1000 R6
SIP call origination & termination
Branch office routing
Proxy to proxy interconnect using ACLs
Least-cost-routing (LCR)
Mixing of PSTN trunks with SIP trunks
TLS support for secure signaling
Route header for flexible call routing through an SBC
Flexible rules for SBC selection (route selection)
Support for Skype for Business SIP trunking

Integration with Microsoft Server 2000 / 2003 and Exchange 2007

Synchronization with Microsoft Active Directory
- Using LDAP interface
- On demand or automatically based on a schedule
- Graphical query design combines ease of use with flexibility
- Allows preview of records to be imported
Dialplan integration with Microsoft Exchange 2007 voicemail server
- Allows mixed environment with groups of users on Exchange or the sipXecs VM server
- Permission based selection of VM server per user or user group
- Automatic dialplan routing to Exchange VM
Enables sll speech based Exchange 2007 capabilities

Supported Softclients

Combined SIP / XMPP clients:
Counterpath Bria professional

Provisioning server for automated mass deployment
Automated SIP and XMPP account setup
Call recording
Supports BLF (workgroups)
Scheduled to support BLA
Automatic user profile and directory management (pending)

Counterpath X-Lite
XMPP clients:
Pidgin
Google Talk
Trillium
Spark
SIP clients:
SIP Communicator
3CX softphone

Analog Lines (FXS)

Supports any SIP compliant FXS gateway
FAX support
Analog cordless phone support
Supports analog Polycom speakerphones
Plug & play management of FXS gateways from Audiocodes, Grandstream and Cisco

Performance

Unlimited number of simultaneous calls (voice, HD voice, video) - only depends on LAN/WAN bandwidth
54,000 BHCC, 120,000 BHCC two-way redundant (depends on server HW)
Up to three-way redundant configuration using cluster mgmt Web GUI
Up to 10,000 users per dual-server HA system
Tested up to 10,000 IM users
450 simultaneous calls through the SIP trunking gateway require < 20% CPU on dual core system
Up to 500 simultaneous conferencing ports per server
Up to 300 media server ports for unified messaging (supports 15,000 users)
Automatic time distribution of re-registration and subscription events

High Availability

Optionally fully redundant call control system
Geo-redundant SIP session manager
Based in DNS SRV (no cluster required)
Load balance under normal operating conditions
Geographic dispersion of redundant systems
Real-time synchronization of state information
Automatic recovery after server failure
Reports on load distribution

Call Detail Records collection and reporting

Call State Events (CSE) collected for all signaling activity
Processing of CSEs into CDRs
All data stored in a database at all times
Flexible report generation using Jasper Reports, built-in
Supports redundant call control
Determines and records call type information
Internal / external calls
Calls to specific sipXecs services
Collates call legs
Historic Call Detail Record reporting in real-time
Additional reports using call type info
Monitoring of currently active (on-going) calls
Export of active and historic CDRs to Excel (.csv file)
Direct database access for reporting application (e.g. Crystal Reports, Jasper Reports)
SOAP Web Services access to CDR data
Individual call history per user in the user portal

Security

All outbound calls authenticated
Secure user password management
DoS attack prevention
HTTPS secure Web access
TLS based signaling for SIP trunks
HTTPS secures non-SIP communication between sipX components.
HTTPS secures communications between sipX components and admin and user consoles.
Secure channel for retrieving messages from voicemail repository.
HTTP digest authentication for SIP signaling, as specified in RFC 2617, is used for authentication challenges between SIP endpoints and sipX components.
HTTP digest implementation supports MD5.

System Administration Features

Browser based configuration and management
Several admin accounts
Notification when new version or patches are available
GUI based software upgrade
GUI based certificate management
LDAP integration
Integration with Microsoft Exchange 2007 for voicemail and Active Directory
SOAP Web Services interface
CSV import and export of user and device data
Administration of Instant Messaging (IM) and Presence settings
Integrated backup & restore
Scheduled backups
Diagnostics
- Display active registrations
- Display job status
- Status of services
- Snapshot logs for debugging
- Logging (customizable log levels, message log per service)
- Display active calls
Domain Aliasing
Support for DNS SRV
Support for DNS NAPTR based call routing
Automatic restart after power failure
- Single sipXecs application can start all other application processes associated with starting up sipXecs, including dependent processes that must be started in particular order.
- Configured from browser interface
Server statistics (integrated graphs and SNMP)
Login history report (successful and unsuccessful)
Automated testing of network services (DHCP, DNS, NTP, TFTP, FTP, HTTP) for proper configuration
Downloadable test tool to run network services tests from a Windows laptop

Plug & Play Device Management

Auto-discovery of phones & gateways on the LAN
Auto-registration of Polycom phones simplifies installation
Plug & play management of phones
Plug & play management of PSTN gateways
Auto-generation of phone / gateway config profile
Auto-pickup of profile by phone / gateway
Centralized management of all the parameters
Centralized backup and restore of all the configs
Auto-generation of lines by assigning users to devices
Device group management & properties
Firmware upgrade management

Unified Messaging (Voicemail)

Integrated unified messaging system
Localized per user by installing language packs
Number of voicemail boxes only limited by disk size (tested up to 10,000)
Performance tested up to 300 simultaneous calls (ports) on dual core server
IMAP back-end connection
Acts as an IMAP client into MSFT Exchange and other compatible email systems
User manageable credentials for IMAP federation
Properly controls MWI on the phone when message is "read" using the email client
Browser based user portal for unified messaging management
RSS feed for new messages
Message Waiting Indication (MWI)
User configurable distribution lists
Group and system distribution lists
Unified Messaging:
Email notification of new voicemail messages
Forwarding of message as .wav file
Supports several parallel notifications
IMAP client into Exchange
Per user selectable templates for email format used when forwarding voicemail
Manage folders: Folders for message organization
Manage greetings: Multiple customizable greetings
Operator escape from anywhere
Remote voicemail access using a phone
SOA Web Services (REST) access to messages and greetings
Unlimited number of inboxes
Auto-removal of deleted messages
Daily report on disk usage sent to admin

Personal Auto Attendant

User configurable personal auto-attendant for every user on the system
Up to 10 individual forwarding choices (keys 0 through 9)
User can record greeting that corresponds with key configuration
Individual zero-out to a personal assistant or receptionist
Individual selection of language based on installed language packs
Personal greeting

Auto Attendant Features

Unlimited number of auto-attendants
Customizable IVR menus with VXML
Dial by extension and name
Night and holiday service
Special auto-attendant
Transfer on invalid response
Nested auto-attendants (multi-level)
Fully customizable actions:
- Operator
- Dial by Name
- Repeat Prompt
- Voicemail login
- Disconnect
- Auto-Attendant
- Goto Extension
- Deposit Voicemail
Uploadable custom prompts
Configurable DTMF handling

Presence Server Features

Centralized presence server based on SIP/SIMPLE
Compatible with Broadsoft or IETF implementations
Centralized management of resource lists for dialog events
Busy Lamp Field (BLF) feature based on presence
Used to support shared lines (BLA)
Presence federated with IM presence to show "on the phone" status
Support for Attendant Consoles
ACD call center agent sign in / out

Hunt Groups

Unlimited number of hunt groups
Serial and parallel forking (rings sequentially or at the same time)
Configurable ring time per attempt
Enable / disable user call forwarding rules while hunting
Flexible configuration of destination if no answer

Call Park Server

Unlimited number of park orbits
Visual indication on the phone of the state of the park orbit using the presence server (BLF)
Music on park
Uploadable music file
Configurable call retrieve code
Configurable call retrieve timeout
Automatic park timeout with configurable time
Configurable park escape key
Allow multiple calls on one orbit

Group Paging Server

Integrated group paging server
Unlimited number of paging groups
Supports regular SIP phones using auto-answer
Supports dedicated in-ceiling devices (SIP)
Configurable paging prefix

Conferencing Server

Voice conferencing server that can run on the same sipXecs server or on dedicated hardware
Support for voice conferencing
Each user on the sipxecs system can have a personal conference bridge
Recording of conference calls
Dynamic conference controls from the user's Web portal (user portal)
Dynamic conference control using IM
Participant entry / exit messages
Roll call
Mute, isolate, disconnect, invite
Association of personal conference bridge with personal group chat room
Automatic migration of group chat to a voice conference using the @conf directive
Support for HD Audio and transcoding if necessary
Support for up to 500 ports of conferencing, dependent on hardware
Configurable DTMF keys for conference controls using the TUI
A sipXecs IP PBX system can have more than one conference server if more capacity is needed
All conferencing servers and services centrally managed and configured
Conferencing based on FreeSWITCH

Call Center Server (ACD)

Supports several ACD servers
ACD server collocated or on a different server hardware
Several (unlimited) queues per server
Several lines per queue
Support trunk lines (many calls per line) or single call per line
Dedicated overflow queues or overflow to hunt group or voicemail
Configurable call routing scheme per queue:
- Ring all
- Circular
- Linear
- Longest idle
Agent barge in (early termination of welcome message if agent becomes available)
Agent presence monitor using presence server
Separate welcome and queue audio
Call termination tone or audio
Configurable answer mode
Agent wrap-up time configurable per queue
Auto sign-out of agents if calls are not answered
Configurable maximum ring delay
Configurable maximum queue length
Configurable maximum wait time until overflow condition
Unlimited number of agents per queue
Statistics:
- Agent statistics
- Call statistics
- Queue statistics
ACD historic reporting (release 3.8)
Supervisor authorization for agent monitoring
ACD historic reports for agents, calls, queues
All reporting stored in database for post-processing if needed

sipXecs Managed Devices

Any SIP compatible phone works with sipXecs if configured manually (i.e. by logging into the phone's Web interface to configure it one phone at a time). The following devices are plug & play managed automatically and centrally by sipXecs:

Polycom SoundPoint all models (IP 301, 320, 330, 430, 450, 501, 550, 560, 601, 650, 670)
Polycom SoundStation IP 4000, 6000, 7000 SIP
Polycom VXX-1500 video phone (release 4.0.2)
Audiocodes gateways MP112, MP114, MP118, MP124 FXS
Audiocodes gateways FXO and PRI/BRI
IPDialog SIPTone V
Counterpath Bria Professional
Avaya 1210, 1220, 1230, 1235 with SIP firmware

sipXecs Managed Devices (experimental support)

Experimental support means that the phone plugin for plug & play management is provided as is. These phone plugins are less frequently updated to the latest firmware and are less tested. Some functionality might not be implemented or supported.

Aastra 53i, 55i, 57i
Snom 300, 320, 360, 370 up to firmware 7.x
Grandstream BudgeTone, HandyTone
Grandstream GXP2000, GXP1200, GXP2010, GXP2020
Grandstream GXV3000 Video Phone
Hitachi IP3000 and IP5000 WiFi phones
Cisco ATA 186/188
Cisco 7960, 7940, 7912, 7905
Cisco 7911, 7941, 7945, 7961, 7965, 7970, 7975
ClearOne MaxIP Conference Phone
LG-Nortel LG 6804, 6812, 6830
Nortel video phone 1535
Linksys ATA 2102, ATA 3102
Linksys SPA8000
Linksys SPA901, SPA921, SPA922, SPA941, SPA942, SPA962
Nortel 1120 / 1140 SIP
G-Tec AQ10x, HL20x, VT20x

Required Hardware

Intel / AMD x86 compatible server
Min RAM 2 GB or more
Linux operating system (RHEL, CentOS or SuSE)
32 bit and 64 bit versions available
PowerPC (PPC) supported on SuSE (need to compile yourself)
No special HW required, sipXecs uses external gateways

Installation and Upgrades

Automated installation from CD ISO for OS and sipXecs IP PBX application
Graphical configuration wizard for system configuration after installation
Certificate generation (allows installing a signed certificate if desired)
GUI based upgrade management from the admin Web interface
Standard Linux package management (e.g. up2date and yum)
Optional auto-configuration of DNS, DHCP, NTP, FTP, TFTP, HTTP servers
Designed so that no Linux admin skills are required for installation and configuration

Centrally Managed sipXecs Distributed System (cluster)

Automated installation and configuration of a distributed system with specific server roles
Automated and central configuration of a high-availability redundant sipXecs system
Allows for dedicated server hardware for conferencing, voicemail, ACD Call Center, and Call Control
All configuration for remote servers is centrally generated and distributed securely

SIP Implementation

This is probably quite an incomplete list. In any case, sipXecs IP PBX is fully SIP standards compliant.

RFC 3261 Session Initiation Protocol using both UDP and TCP transports
Advanced call control using RFCs
- RFC 3515 Refer Method
- RFC 3891 Referred-By header
- RFC 3892 Replaces header
Provide for consultative and blind transfer and third party call controls
- Blind transfer (Unannounced) to a different phone without speaking to the other phone prior to transfer.
- Consultative transfer (announced) to a different phone without speaking to the other phone prior to transfer.
- Consultative transfer (announced) to a different phone after speaking to the other phone prior to completing the transfer. (Consultative transfers require a SIP phone that supports this feature)
RFC 3263 Locating SIP Servers - use of DNS SRV records for call routing control and server redundancy.
RFC 3581 Symmetric Response Routing (rport)
RFC 3265 SIP Event Notification - for phone configuration and
RFC 3842 Voice mail message waiting indication (MWI)
RFC 3262 Reliable Provisional Responses
RFC 2833 Out-of-band DTMF tones
RFC 3264 Offer/Answer model for SDP for Codec Negotiation
RFC 2617 HTTP Authentication: Basic and Digest Access Authentication
RFC 3327 Path header
RFC 3325 P-Asserted identity
RFC 4235 An INVITE-Initiated Dialog Event Package for the Session Initiation Protocol (SIP)
RFC 4662 A Session Initiation Protocol (SIP) Event Notification Extension for Resource Lists
RFC 2327 SDP: Session Description Protocol
RFC 3326 The Reason Header Field for the Session Initiation Protocol (SIP)
Early media (SDP in 180/183)
Delayed SDP (SDP in ACK)
Re-INVITE: Codec change, hold, off-hold
Route/Record-Route header fields
Configurable RTP/RTCP ports
Configurable SIP ports
BLA support

RFC 3680: A Session Initiation Protocol (SIP) Event Package for Registrations
RFC 3265: Session Initiation Protocol (SIP)-Specific Event Notification
draft-ietf-sipping-dialog-package-06
draft-anil-sipping-bla-02

XMPP Compliance

RFC 3920: XMPP Core
RFC 3921: XMPP IM
XEP-0030: Service Discovery
XEP-0077: In-Band Registration
XEP-0078: Non-SASL Authentication
XEP-0086: Error Condition Mappings
XEP-0073: Basic IM Protocol Suite
XEP-0004: Data Forms
XEP-0045: Multi-User Chat
XEP-0047: In-Band Bytestreams
XEP-0065: SOCKS5 Bytestreams
XEP-0071: XHTML-IM
XEP-0096: File Transfer
XEP-0115: Entity Capabilities
XEP-0004: Data Forms
XEP-0012: Last Activity
XEP-0013: Flexible Offline Message Retrieval
XEP-0030: Service Discovery
XEP-0033: Extended Stanza Addressing
XEP-0045: Multi-User Chat
XEP-0049: Private XML Storage
XEP-0050: Ad-Hoc Commands
XEP-0054: vcard-temp
XEP-0055: Jabber Search
XEP-0059: Result Set Management
XEP-0060: Publish-Subscribe
XEP-0065: SOCKS5 Bytestreams
XEP-0077: In-Band Registration
XEP-0078: Non-SASL Authentication
XEP-0082: Jabber Date and Time Profiles
XEP-0086: Error Condition Mappings
XEP-0090: Entity Time
XEP-0091: Delayed Delivery
XEP-0092: Software Version
XEP-0096: File Transfer
XEP-0106: JID Escaping
XEP-0114: Jabber Component Protocol
XEP-0115: Entity Capabilities
XEP-0124: HTTP Binding
XEP-0128: Service Discovery Extensions
XEP-0138: Stream Compression
XEP-0163: Personal Eventing via Pubsub
XEP-0175: Best Practices for Use of SASL ANONYMOUS

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