Supported platforms:

Join for Mobiles is a set of feature-rich VoIP clients developed by VoipSwitch for the most popular mobile platforms. The clients are based on SIP and other open standard technologies supported by IMS and VoLTE. Its advanced underlying qualities complimented by the sleek design and the deep understanding of User Experience is what makes Join the market leader.

Join is a base for building a white label fully customized communication application.

For functionalities like group chat, Enhanced Address Book, file transfer, see Rich Communication Suite clients.

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• Short time to market Complete client framework reduces development, integration and testing phase.

• Interoperability: Join is standard based and has been tested with all popular softswitches and IMSes in the market – Huwei, ZTE IMS, Broadosft. Nokia Siemens Network, PortaOne and many others

• Unparalleled User Experience: Join utilizes advanced mechanism for NAT and Firewall passing through. It supports ICE with STUN/TURN and TCP TURN. It just works from any network.

• HD audio and Video codecs, compatible with WebRTC

• Works with 3^rd party SIP softswitches and IMSes (tested with Digitalk, Portaone, MediaCore, Mera, Aloe, Broadvox, Samsung IMS, Nokia Siemens Network IMS, ZET IMS and others)

• Multiple accounts support with multiple active registrations (up to 20 accounts)

• Background support for TCP, UDP, TLS and PUSH (in connection with Always On server)

• Works over cellurar data (3G) or WIFI

• Compatible with iPhone, iPad and iPod

• Built-in tunnel* for passing through VoIP blockades

• Audio codecs: OPUS, G.711, G.722 (NB, WB), G.729, GSM, Silk (NB, MB, HD, UWB), Speex, AMR NB and WB, Opus (more on request)

• Video codecs include: H264, H263+, H263, VP8

• Echo Cancellation, Noise reduction

• Peer to Peer media path (even for clients behind NATs) using ICE technology

• Preprocessing algorithms including Echo Cancellation, Noise suppression

• WebRTC media engine (Android)

• Resolutions from CIF to HD (subject to device’s performance), Frame rate up to 30 fps

• Congestion control

• Multiple calls support – both for audio and video calls

• Switching between active calls

• Call conference (up to 5 parties, mixing on the device side)

• Call recording

• Call transfer (blind, attended), hold

• Speaker, mute

• Video Call preview

• Switching between front and back camera

• Balance information

• Destination, rate information

• Call history logs

• Full SIP compliant

• IMS/VoLTE extensions support

• Presence with Resource List, Watcher Info

• XCAP for XDM

• STUN/TURN and ICE framework (peer to peer and NAT traversal)

• Instant messaging using SIP SIMPLE (OMA), CPM, CPIM standard

• Presence: centralized model, Watcher Info, Resource List

• Presence integrated with contacts

• XCAP support for XDM servers

• Conversational view of messages (chats)

• IM large messages (MSRP)

• Deferred messages

• SMS over IP (to OFFNET recipients)

• Converged inbox

• SRTP encryption for audio and video

• AES with 128 or 256 bit key

• TLS encryption for signaling

• zRTP

• GSMA IR.92 & IR.94 specification support

• IMS Registration, Authentication and addressing

• VoLTE Voice and Video call establishment

• IMS Preconditions

• Voipswitch RCS extensions support

• GSMA RCS standard compliant

• User Discovery using Options, File transfer, Provisioning

• Dialing plan support – create your own dialing plan rules

• International dialing – automatically add prefixes to dialed numbers

• Contact list integrated with native address book

• Favorites contacts

• Detailed call history

• Support for VPN

• Bluetooth and headset support

• Voicemail indicator (MWI)

• Visual voicemail (in connection with VoipSwitch)

• syncML OMA address book support

• Call through (calling through a predefined access number)

• Callback (triggering a call through http callback request)

• XMPP (jabber)

• Social networks integration (full chat and presence for Gtalk and Facebook)

• T140 support for real time messaging