Test App

VoIP calls with auto QoS using the SIP protocol standards (both incoming and outgoing calls)
Connects directly to your preferred VoIP server (no need for any intermediary media servers)
Android OS: all versions from 2.0
Connection: WiFi or mobile GPRS, EDGE, 3G, 4G (all above 15 kbits)
Transport protocols: UDP, TCP, TLS, HTTP, VPN (with VoIP tunneling and encryption)
Built-in tunneling and encryption (for both the signaling and the media)
Peer to peer encrypted media
Audio Codec: G.711 (PCMU and PCMA), GSM, Speex, iLBC, G.729 (G.729 is restricted in the free version), HD Audio/wideband
Video Codec: H.264, H.263 (beta)
Automatic fine-tuning (including codec selection) depending on your peer capabilities, network speed and CPU power
AEC (acoustic echo canceller), Denoise filter, AGC (automatic gain control), PLC (packet loss concealment), Silence suppression
Multiple accounts
Detailed or simple configuration
System phone-book
Call history
Call recording
Balance and rating display, call timer, status logs, detailed logs
Multi-tasking support -listening for incoming calls in the background
Speakerphone, Mute and Hold
IM (chat), SMS
DTMF (RFC2833 or INFO method)
NAT/Firewall support, stable SIP and RTP ports, auto QoS, light STUN protocol and auto configuration
Call park and pickup
Call transfer (attended and unattended) and call forward
Conference calls (built-in RTP mixer and transcoding when necessary)
DNS SRV record lookups
Supported methods: INVITE , ACK, PRACK, BYE, CANCEL, UPDATE, MESSAGE, INFO, OPTIONS, SUBSCRIBE, NOTIFY, REFER
RFC’s: 2543, 3261, 2976, 3892, 2778, 2779, 3428, 3265, 3515, 3311, 3911, 3581, 3842, 1889, 2327, 3550, 3960, 4028, 3824, 3966, 2663, 3022 and other drafts
signaling and media engine