PROMO * MVTS II v.1.3.1-50 to 1.4.0-50 - Professional Installation and Consulting for Setup - Training *

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 =93hosted softswitch=94 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

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VoIP SIP SDK:

=95 g729 and g723 Codec=B4s support =95 Multiple and single Codec selection support =95 Failure codes support (get SIP Message Response Code, SIP Message Response Text) =95 RTP/RTCP Port setting (for inbound RTP traffic) =95 Reduce audio latency and audio latency settings (properties: MinPrefetchCount, MaxPrefetchCount, MaxRTPPackets) =95 Media status (Events: OnLocalMediaStarted, OnLocalMediaStoped, OnRemoteMediaStarted, OnRemoteMediaStoped) =95 Get used codec per line =95 Custom Ringtone (play wav) support (property: RingtoneFile) =95 Play wav to a selected phone line (methods: StartPlayingAtLine, StopPlayingAtLine) =95 Redirect Call to other phone line =95 Load and Save Configurations (methods: LoadConfiguration, StoreConfiguration) =95 Complete new, re-written and updated samples with source code =95 and much more!

Here is a list of the main features of the VoIP SIP SDK::

=95 Easily make and receive SIP (Session Initiation Protocol) based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider =95 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 =95 Open standards-based and interoperable with all of the major equipment vendors =95 UDP and TCP support =95 Multi-party voice conference support/ Conference split and join, locally mixed conferences =95 Multi-line support (multiple simultaneous calls) =95 SIP Instant/Chat Messaging with send/receive controlling =95 Integrated STUN, TURN and ICE support

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Max Loger

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