The Grandi GIP300 IP phone is a new entry level IP phone. GIP300 has dual 10/100M Ethernet ports that can be configured as either switched or routed ports, built-in NATi router/DHCP server, high quality full duplex hands-free speakerphone with acoustic echo cancellation, headset jack, message waiting indicator, and more memory for future function growth. It supports G.711i, G.723.1i, G.729A/B. GIP300 is able customer to register two SIP servers simultaneously. In addition, you could download music ringing tones and the GIP300 provides as many as eight kinds of ring tongs.
The Grandi GIX100 VoIPi ATAs works great. It is very easy to configure, excellent audio and great selection of codecs including G.711, G.723.1, G.729A/B. It is inexpensive with dual 10/100M Ethernet ports that can be configured as either switched or routed ports, built-in NAT router/DHCP server and support message waiting indicator. And most importantly, GIX100 is an ultra-affordable VoIP ATAs with high performance.
Both GIP300 and GIX100 is a new generation VOIP terminal products which were developed based on single ARM9E CPU platform. Its core is a ARM9E CPU at 150MHZ. With this technology, Grandi would provide the products with high performance and ultra-affordability cost.
GIP300 is just at $30.00 USD for 100K units order. GIX100 is just at $28.00 USD for 100K units order.
TERRATELprovided SIP/E1 Gateway, a digital TDM switch supporting SIP (compatibility with the Asterisk, FreeSWITCH, SoftSwitch etc.), is a universal product designed to provide VoIPi and TDM telephony solutions. The switch is aimed at optimizing solutions for building and upgrading corporate networks by applying open Asterisk/FreeSWITCH/SoftSwitch -based IP telephony solutions. The figure shows the block diagram demonstrating the possible application of the switch.
Features: 1. SIP Voice and Video call 2. STUN support for NATi traversal 3. Multiple codec selection and ordering of codecs 4. Blind call transfer 5. Call transfer with consultation (attended call transfer) 6. Call reject 7. Call Hold 8. System tray icon
SIP Client SDK provides a powerful and highly versatile solution to add quickly SIP (Session Initiation Protocol) based dial and receive phone calls features in your software applications and websites. It accelerates the development of SIP/RTPi compliant soft phone with a fully-customizable user interface and brand name.
The VoIPi SIP Client SDK contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users and SIP compatible devices (hardware and software). It enables a worldwide communication over the internet or intern networks either and delivers superior voice quality by integrating digital voice processing features including auto gain controller (AGCi), acoustic echo suppression (AES) and noise suppression. It supports multiple lines, multi-party voice conference, call hold, call forwarding and transfer, DTMFi, Packet Loss Concealment (PLC), adaptive jitter buffer, record and playing WAV and much more!
VoIP SIP Client SDK is based on IETFi standards (SIP, RTP, STUN, TURN, ICE etc.), so it should be compatible with other standard based products such as: SERi, Sip EXpress, OpenSER and Asterisk.
New features of the VoIP SIP Client SDK:
• g729 and g723 Codeci´s support • Multiple and single Codec selection support • Failure codes support (get SIP Message Response Code, SIP Message Response Text) • RTP/RTCPi Port setting (for inbound RTP traffic) • Reduce audio latency and audio latency settings (properties: MinPrefetchCount, MaxPrefetchCount, MaxRTPPackets) • Media status (Events: OnLocalMediaStarted, OnLocalMediaStoped, OnRemoteMediaStarted, OnRemoteMediaStoped) • Get used codec per line • Custom Ringtone (play wav) support (property: RingtoneFile) • Play wav to a selected phone line (methods: StartPlayingAtLine, StopPlayingAtLine) • Redirect Call to other phone line • Load and Save Configurations (methods: LoadConfiguration, StoreConfiguration) • Complete new, re-written and updated samples with source code • and much more!
Here is a list of the main features of the VoIP SIP Client SDK:
Easy, familiar, event-driven call control ActiveX • Easy to use; quick development • Support for .NET framework and all development environments with ActiveX support • Very easy to incorporate
Rich call control feature set • Multi-party voice conference support (Conference split/ join, locally mixed conferences) • Multi-line support (multiple simultaneous calls) • SIP Instant messaging • Locally mixed conferences • Hold/Mute • Call transfer • Call forwarding and rejection
Industry leading SIP support • RFC3261 compliant SIP stack • RFC 2833 out-of-band DTMF signaling • Integrated STUN, TURN and ICE support
Comprehensive configuration support • Select media input/output devices (on-the-fly as well during a conversation/conference) • Configurable ports (RTP, SIP UDP, SIP TCP, STUN, TURN, ICE) • SIP proxy
Advanced digital voice processing features • AGC (auto gain controller) • AES (Acoustic echo cancellation or suppression) • Noise cancellation or suppression
A set of RFC-compliant high-functionality class libraries which manage SIP messaging. Designed to support the development of cross-platform multimedia and text-based applications and services written any any .NET language (e.g. c#, VB or F#).
◆ SIP (RFC3261, RFC2543) ◆ Support IAX2 protocol ◆ Support codec: G.711A/u, G.722,G.723 high/low, G.729i A/B ◆ Support G.168 echo cancellation standard, compliant 96ms echo cancellation with speaker mode ◆ Support voice volume adjustment, including IN/OUT of handset and speaker ◆ Support Jitter Buffer, VAD, CNG, SIP, Domain name register, point-to-point Call ◆ Support RTPi and RTCPi ◆ Support the Inbound/Outbound transmission; SIP info, DTMFi Relay, RFC2833 ◆ Support many countries' standard ring ◆ Support NATi: Support STUN, CITRON, AVS Mode ◆ Support SIP domain, SIP Authentication (none, basic, MD5), Domain Name parse ◆ Support two SIP server synchronously, including Pubic Server/ Private server, can make a call by any proxy. You can back-up and select any above SIP server. ◆Support SIP application, including SIP Call forward/transfer/holding/conference/pickup/redial/unredial/joincall
I'm Sophia from Wuchuan Network (Shenzhen) Ltd. Info below is about our best sale product for now---SIP/IAX2 VoIPi Phone PH802.which is a unique SIP Phone featuring for IP PBXi.Its brand is Wuchuan or Neutral. If you are interested in our product, pls don't hesitate to contact me.
I'm Sophia from Wuchuan Network (Shenzhen) Ltd. The following Info is about our newly high tech product --SIP/IAX2 VoIPi Phone PH806 -- A newly developed hi-tech SIP phone for enterprise supporting 5 SIP servers synchronously,(POE) function and 128*64 LCD display.Besides, it supports 3 interactive soft key to play humanized operating prompt and an option of a pair of headsets for private calls.
Our PH806 has all basic functions that other IP phones have, and belows are our phone’s special and advanced functions:
IP Phone PH217 completely follows VOIP standard offered by ISO, setting in two Protocols: SIP, fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software in the market.Simple IP phone for home users
The two IP Phones are our best sale products for now. The unique phones can display directly the online state of the booking numbers by the indicator. If the indicator is green, means it is online; the indicator is green and twinkle, means it is in the course of the call; if red, means it is offline.