SIPez's End to End HD Voice and Wideband Audio Solutions
We have software solutions for your embedded hardware product as well as
software application or server based products. Our solutions are very modular
allowing you to incorporate only the capabilities you need to keep a small
code footprint and minimal resource requirements. We have a OS independence
layer that allows our solution to work on your operation system. We currently
support most operating systems including: Linux, VxWorks, Windows,
Mac OS X, WinCE, Solaris, Symbian and Android.
Android SIP Client Applications
SIPez can provide off-the-shelf or custom built Android SIP applications to fit your needs. Android SIP client application product sheet
Web Browser SIP Client
SIPez's ezF2SIP Flash phone enables SIP calls from standard web browsers on Windows, Mac OS X and Linux. See our ezF2SIP product sheet for more details.
Wideband and HD Voice Conference Bridge, IVR Engine and Applications
SIPez's ezIVR interactive voice response engine supports wideband and HD voice audio for the clearest quality and best communication possible for your conferencing and IVR applications. Use our standard IVR applications or let us customize our conference and IVR applicatiions to satisfy your needs. We can provide small scale single box solutions all the way up to large scale multi-host high availability solutions. See the ezIVR product sheet for details.
- Media Processing Engine
- Adaptive Dejitter
- Packet Loss Concealment
- Voice Activity Detection
- Active Gain Control
- Acoustic Echo Cancelation
- Conference Bridge Mixing
- Speaker Selection
- Side-bar Conversations
- Whisper Mixing
- Plugable Codecs
- Audio File Playout
- Audio Buffer Record and Playout
- Selective Recording
SIPez has a full featured audio processing subsystem
including a RTP and RTCP stack. The audio processing subsystem
supports plugin codecs, audio file streaming, echo cancelation,
bridge mixer, stream recording,
adaptive dejitter, lost packet reconstruction. The combined
solution creates extremely high quality audio capabilities
even in adverse network environments. Our audio processing
solutions are suitable for embedded systems running on pure
host processing or hybrid host processor and DSP platforms
as well as pure software based solutions running standard
operating systems. Our audio processing subsystem is used
in commercial products from small scale embedded devices
such as SIP phones to large scale products such as media
servers and gateways.
Test Network Simulator
SIPez's Cumulus Network Simulator and tools provide the ablity to
generate, capture and reproduce adverse network situations in which
packet loss and delay diminish audio quality and reliablity.
This tool is extremely useful for testing the effectiveness
of audio quality improvements enabled by adaptive dejitter
and packet loss compensation. SIPez can provide the tools or
complete testing service of your product's performance under
typical network behavior.
SIPez has solutions to power next generation SIP hosted voice
providers. If combined with our consulting services, we can provide a
turn key solution.
Media & Conference Servers
We can provide enterprise class media servers to handle your interactive
voice response (IVR) needs. Depending on requirements, solutions range
from fixed implementations to open and flexible vxml based scripts.
SIP conference bridge functionality is also available. Please contact
us with your port density needs.
SIPez can help you leverage ready-made and tested
PBX solutions for both server and embedded environments. We
have experience building, customizing, and deploying solutions
on embedded systems, Linux, Solaris, and Microsoft Windows.
Phones (Appliance & PC client) & Gateways
We have proven solutions for SIP end points. Whether you are
building hard phone appliances, soft phones, SIP to PSTN
gateways, terminal adaptors, media and IVR servers or
conference bridges, we have the software components and complete
solutions you need to rapidly develop and deploy your product.
Our solutions are well tested and provide you with the highest
level of interoperablity availble on the market. Because we
leverage open source our solutions are also the most economical
as well. By reducing the cost of the core VoIP functionality,
we enable you to focus your efforts on your value added product
differentiation. Our solutions include capabilities for:
- Call Processing
- Media Processing
- Conference Bridging
- REFER based blind and consultative transfer
- STUN and ICE based NAT traversal
- Third Party Call Control
Presense Clients & Servers
SIPez has fully functional SIP Proxies, Registrar and
Redirect servers that are highly configurable as is. These servers
are also easily extended to provide applicaition specific
features through plugin APIs. SIPez can customize
these servers or repackage them for OEM solutions. These servers
have been tested and proven to be RFC 3261 compliant and highly
interablerable through SIPit and other test events.
SIPez has experience with and can provide open source SIP stacks such as
JAIN SIP, reSIProcate, and sipX stack for your platform. We can help you understand
how to apply these SIP stacks or your preferred commercial stack.
We understand the capabilities and weakness of the leading SIP
stacks and can help you decide which is best for your application.
We can also provide design and development skills to implement your
required SIP capabilies for your product or application.