Discussion:
PROMO * MVTS II v.1.3.1-50 to 1.4.0-50 - Professional Installation and Consulting for Setup - Training *
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softswitch solutions
2010-09-03 14:31:52 UTC
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MERA VoIP Transit Softswitch II (MVTS II) is a next-generation prepaid
switching platform with a geographically distributed architecture and
highly flexible traffic handling capability. MVTS II is specifically
designed to increase the efficiency of VoIP traffic management on
large-scale networks and is targeted towards carriers running 3+
million minutes of VoIP calls per month. Built on modular
architecture, the versatile MERA solution alleviates the challenges of
managing traffic flows on highly distributed networks by providing
intelligent built-in algorithms of call routing and elaborate analysis
and reporting tools. MVTS II is a handy solution for wholesale
carriers who consider price-to-quality ratio in each call route and
need to promptly react to changes in routing policy of their peers.
MVTS II is intended for carriers that would like to keep all billing
and routing data in one single database (powered by Oracle), included
into the system. MVTS II is ideal for carriers that would like to take
advantage “hosted softswitch” capability and sacrifice ease of the
systen management for elaborate functionality.


The rich feature set of the MERA solution includes:

* New intelligent routing capabilities
* Computer-aided profitability monitoring
* Native support of SIP and H.323
* Various proxy options for both SIP and H.323 (Two-way SIP/H.323
conversion)
* Codec conversion (G.729, G.729A, G.723.1, G711A-Law, G.711mU-
Law, GSM FR, Speex, iLBC)
* Load balancing mechanism
* Elaborate analysis and reporting tools
* Advanced pre-billing and accounting capabilities
* Partitioning capabilities (Hierarchial system partitioning)
* Number portability support
* Prepaid features
* Inter/Intrastate routing
* Real-time profitability control
* Billing (included)
* Call Statistics and Analysis (QoS, ASR, ACD etc.)
* Load balancing mechanism, cluster and replication system is
possible

Minimal System Requirements

* Depend on the system configurations and required capacity

Capacity

* Up to 20,000 concurrent calls
* Call accretion rate (CPS): 250
* New registrations per second: up to 100

Minimal configuration capacity (2 servers):

* Up to 1,000 concurrent calls (without codec conversion)

Signaling node capacity:

* Up to 3000 simultaneous calls


- Multi systems - load balacing - cluster setup *
- Software Upgrade, capacity upgrades , database problems , technical
support , Training , available

We provide full installation, at some extra fee you can order tech
support, configuration of your system per your requirements.
We can do updates for you after Mera releases the new version with
some few update fees Affordable prices,
quick problem resolving, professional attitude!


You can contact me via email or msn : mera2solutions (@) hotmail.com
to ask any questions!
Gmail: mera2solutions (@) gmail.com
Max Loger
2010-09-06 06:37:28 UTC
Permalink
http://ipphonesdk.com

VoIP SIP SDK:

• g729 and g723 Codec´s support
• Multiple and single Codec selection support
• Failure codes support (get SIP Message Response Code, SIP Message
Response Text)
• RTP/RTCP 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 SDK::

• Easily make and receive SIP (Session Initiation Protocol) based
phone calls through any SIP gateway or SIP compliant IP-Telephony
service provider
• VoIP conferencing with crystal clear sound even for both low and
high-bandwidth users
G711 A-Law, G711 U-Law, Speex, Speex-wb, GSM6.10, iLBC, L16 and
g729 & g723 Codec
• Open standards-based and interoperable with all of the major
equipment vendors
• UDP and TCP support
• Multi-party voice conference support/ Conference split and join,
locally mixed conferences
• Multi-line support (multiple simultaneous calls)
• SIP Instant/Chat Messaging with send/receive controlling
• Integrated STUN, TURN and ICE support
<• Comes with new sample SIP Proxy Server to provide in bundle with
the SIP Client ActiveX a ready up own SIP VoIP and Instant
Messaging network solution.
• P2P support for directly connections between 2 SIP clients
without SIP Server
• Outbound proxy server support
• Encrypted SIP account settings (encrypted SIP account settings in
your webpage)
• Line Hold/Un-hold support
• Call forwarding and rejection
• Call transfer support
• Select media input/output devices
on-the-fly - also during a conversation/ conference)
• Mute microphone/speaker + level indicator
• Auto-answer
• DND (Do Not Disturb)
• Adaptive Jitter buffer
• PLC (Packet Lost Concealment)
• AGC (auto gain controller)
• AES (Acoustic echo cancellation or suppression)
• Noise cancellation or suppression
• DTMF tones support (generation/detection)
• Recording voice conversation into PCM WAVE (.wav) file
• Playing PCM WAVE (.wav) files to the remote end
• Audio file memory cache
• Extended SIP URL functions
• Dynamically loadable codec support (coming soon)
• Comes as ActiveX control (Web demo with ready-up signed CAB
included)
• Registration on SIP Server (SIP Registrar)
• Log file on/off setting
• Microphone and Speaker Volume with Mute support
• Keep-alive packets to NAT/firewall
• Fully-customizable user interface
• Microsoft Authenticode Certificate
• Works with all kind of Internet connections
• Friendly to NAT and other firewalls
• Keep-alive packets to NAT/firewall
• Royalty free licensing
• No Yearly/Monthly fee
• Very easy to incorporate
• Fully sample applications for various programming languages such as
sample source code for C#, VB.NET, JavaScript (Webdemo), VB 6.0 and
Delphi
• For .NET framework as well and all development environments with
ActiveX support

Easy, familiar, event-driven call control ActiveX of VoIP SIP SDK:
• 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 of VoIP SIP SDK:
• 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 of VoIP SIP SDK:
• RFC3261 compliant SIP stack
• RFC 2833 out-of-band DTMF signaling
• Integrated STUN, TURN and ICE support

Comprehensive configuration support of VoIP SIP SDK:
• 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 of VoIP SIP SDK:
• AGC (auto gain controller)
• AES (Acoustic echo cancellation or suppression)
• Noise cancellation or suppression

… VoIP SIP SDK supports even much more!

http://ipphonesdk.com

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