Dspic ilbc jobs
need softphone app for iphone. i need similar app for android and windows and mac desktops from the basic futures of a softphone , it must put voip call on hold when gsm call occurs. It must support codecs like g711(alaw,ulaw, speak,iLBC, GSM, G729, h263+,vp8, h264*.
...the micro controller PIC18F46J50 from Microchip. The result should be displayed in a terminal program on the PC. Attachment: Similar project from the mTouch Developing kit from Microchip. The target can be reached also by using the attached project and reorganize and simplify it, so that it will fit exact to the target: Project "Basic Button Demo" in subdirectory ctmu1/mTouchCapDemos/PIC18F--dsPIC-Demos/BasicButtonDemo, - reorganizing the subdirectories, for example , in the subdirectories "Microchip/include/mTouchCap" and "Microchip/mTouchCap" some files are doubled, rename the variables and functions to fit better to the target, etc. - excluding the unnecessary options from the code, for example using different compilers, different b...
Hello, I need a class to be written using AudioUnit/AudioToolbox that will be able to read data from iPhone's microphone, in realtime convert the data to AAC or iLBC, and store that data in the buffer. Class must work both ways, which means it also needs to read the incoming buffer, decode from AAC or iLBC, and write it to iPhones speaker. I need it urgently. I can pay 50-70$.
Hello, I have a class that reads input from iPhone recording device and sends it to singleton. Also this class read data from singleton and writes it to iPhone playback device. What I need to do is to create method for converting recorded buffer from PCM to AAC or iLBC format, and later to decode this AAC or iLBC back to PCM. It's done using AudioConverterFillComplexBuffer. It's really a few lines of code. All existing code is provided. This is needed for TODAY! I offer 50-70$ for working code.
We are looking for a talented android(and possibly iphone) developer that is experienced in VOIP. Skills required: - sip (for audio and video calls) - xmpp (for presence and messaging) We would like a person who has experience with using a sip stack library for making audio and video calls (H264 for video, ilbc, opus or similar for audio) and using an xmpp library for mobile phones (mainly android but iphone is also a big plus). Also that person should know how to use NAT traversal technologies like ICE protocol, STUN and TURN servers. Also big plus is knowledge of adaptive bit rate and other means of changing video and sound quality based on the quality of internet connection. Before answering for the ad write "I CAN DO THIS" as the first li...
...Identification Presentation (CLIP) ?Calling Line Identification Restriction (CLIR) ?Call Forward All ?Call Forward Busy ?Call Forward No Answer -Sound quality must be High quality, if possible HD sound quality, -Beside good sound quality minimum bandwidth rate must be used at particular connectivity. o Echo cancellation is preffered o It should support a wide variety of audio codecs like iLBC, PCMA, PCMU, g.723, g.729... -Address book integration, and call history (Dialed/Answered/Missed). -Branding is mandatory – skinning, -Custom selection of 3G or WiFi data connection; o When WiFi network is connected, SIP caller application starts automatically (if user set this option). 2. Second step, Phase 2: Simple VoIP application - SIP caller with pro...
SIP Mobile Dialer (iPhone, Android and PC) with VPN Tunnel: It Should work from any block or unblock country as UAE, OMAN, KSA etc. Customized design with our company logo. Use SIP and H.323 for signaling. Compatible with all Soft Switches, supporting SIP. User friendly inter...Customized design with our company logo. Use SIP and H.323 for signaling. Compatible with all Soft Switches, supporting SIP. User friendly interface. Flexible integration with your Phone Book contacts. Support Incoming & Outgoing call. QoS for VoIP Calls. Display Balance. No extra Balance server is required. Dialed/Received Call History. Call timer. Refill Account. Support G729, AMR, ILBC, GSM, Speex, G711 codec Compatible with EDGE, GPRS, Wi-Fi & VPN Access points NAT Traversal or ...
...our softswitch with unlimited user licence. requirements: Call Recording Call Conference video call Hold / Unhold Transfer/forward to mobile or landline DND (Do not Disturb) Redial Mute Auto accept call Codec Supported G729,G723, G711 u, G711 a, G722, GSM, iLbc, Speex/ 8000, Speex/16000 , Speex / 32000 Codec selection and Codec Quality Control (Bandwidth control) Silence Suppression Echo Cancellation Uses NEW RFC 3261 compliant stack DTMF (RFC 2833) RFC 3951: Internet Low Bit Rate Codec (iLBC) Proxy Settings Registration Timeout Acoustic Echo Cancellation. Packet concealing Packet Lost Concealment (PLC). Comfort Noise Generator (CNG), Resampling Balance Display Credit time display: Customized skin interfaces Call timer Last Numb...
...clients for third party softphones and devices - Voip clients size select features - Sip protocol stack and related protocols - H323 protocol stack (gateway and keeper) - Class4 features (transit, routing, billing, LCR, BRS, load balancing, failovering) - Class5 features (PBX features, ACD, call forward, transfer, DID, etc) - Codec support: all (G.729, G.723, G.711, G.722, G726, GSM, speex, iLBC, wideband, ultra-wideband, H264, DTMF, etc) - Video, Presence, IM support (SIP SIMPLE) IVR Basic calling cards (requires IVR basic) - P2P, callback (requires IVR basic) - IVR Advanced - SMS out - SMS in - Enduser webportal - Resellers - Callshops Tunneling and encryption - Recharge, Paypal, pincodes - Payment Gateways - GeoIP (for location based services) - Dynamic-autom...
VOIP software for: iOS and Andriod SIP Client Phone Phone must support Audio Code G.727, G.711, iLBC, G.722 sample software as in these websites:
I am looking for Website for VOIP Service Provider with following feathers, but in next phase im also looking for Smart Phone App. 1) Website Professional Interactive Website (Web 2.0) 8 - 10 Static pages Customized CMS Sliding / Flash banner Dynamic page for calling rate (as website) Contact Us page. Sample / Concept Website http://www.re...Interactive Website (Web 2.0) 8 - 10 Static pages Customized CMS Sliding / Flash banner Dynamic page for calling rate (as website) Contact Us page. Sample / Concept Website 2) Smart Phone App: Software for Smartphone as well. Operating System : iOS and Andriod SIP Client Phone Phone must support Audio Code G.727, G.711, iLBC, G.722
I am looking for Website for VOIP Service Provider with following feathers, but in next phase im also looking for Smart Phone App. 1) Website Professional Interactive Website (Web 2.0) 8 - 10 Static pages Customized CMS Sliding / Flash banner Dynamic page for calling rate (as website) Contact Us page. Sample / Concept Website http://www.re...Interactive Website (Web 2.0) 8 - 10 Static pages Customized CMS Sliding / Flash banner Dynamic page for calling rate (as website) Contact Us page. Sample / Concept Website 2) Smart Phone App: Software for Smartphone as well. Operating System : iOS and Andriod SIP Client Phone Phone must support Audio Code G.727, G.711, iLBC, G.722
...voice recording into ilbc compression ( ) - push the voice file to the server using json & ftp (to be provided when selected) - an open port to receive push notification from the server via json to download new messages and also trigger UI elements (instructions to be provided when selected) - login to facebook & grab user pictures & facebook id & name using facebook api () The objective is to create the backend code frame base for use to work with the UI elements - AS3. Outcome wanted is source code for: 1) do a simple facebook login & display the user's image, name & facebook id 2) a simple flash button to trigger a voice recording for 10 sec 3) voice to save in iLBC compression 4) upload
...recording into ilbc compression (<>) - push the voice file to the server using json & ftp (to be provided when selected) - an open port to receive push notification from the server via json to download new messages and also trigger UI elements (instructions to be provided when selected) - login to facebook & grab user pictures & facebook id & name using facebook api (<>) The objective is to create the backend code frame base for use to work with the UI elements - AS3. Outcome wanted is source code for: 1) do a simple facebook login & display the user's image, name & facebook id 2) a simple flash button to trigger a voice recording for 10 sec 3) voice to save in iLBC compression 4)
...TCP, TLS 4. NAT/Firewall Support: stable SIP and RTP ports, rport support, STUN protocol and auto configuration. 5. QOS Support for TOS byte for voice and video 6. Provide capability to select audio device in, audio device out, video device in, video device out 7. The vendor should provide the name of the codec libraries used in the solution. 8. Voice Codecs: G711A, G711U, iLBC, the vendor should also provide cost of adding codecs in the future for voice and video codecs. 9. Voice Features: a. PLC - Packet Loss Concealment b. AEC - Acoustic Echo canceller c. Noise Suppression d. Silence Suppression e. AGC - Automatic Gain Control f. Video Mute g. Setting Volume h. Jitter Buffer - Provide depth and algorithm 10. Vide...
...against brute force and other attacks highly stable and reliable below specs: # VoIP calls with auto QoS using the SIP protocol standards (both incoming and outgoing calls) # Android OS: - and above # Connection: GPRS, EDGE, WiFi, 3G, 4G (all above 15 kbits) # Built-in tunneling and encryption (for both the signaling and the media) # Codec: G.711 (PCMU and PCMA), GSM, Speex, iLBC, G.729 G.723 # 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 # System phone-book # Call history # Balance and rating display, call timer, status logs, detailed logs # Multi-tasking supp...
# About the Project: ? We have a PIC based microcontroller board and we need a software to be developed to operate this board (we actually have about 500 of them) ? The device will act like an AMR or remote telemetry board ? The purpose is: to collect some data from nearby devices via RS232/RS485, log them on internal EEPROM and send the log to the server periodically while contacting the server immediately in special cases (alarms - based on collected value) # About the Hardware: ? The board uses a dsPIC30f6014A controller ? The board has a 45DB161D EEPROM ? has a QuacTel M12 GSM module ? Some of the IO pins are wired to external connector to be used as GPIOs ? has an RS485 port as well as 6 RS232 ports (1 as debug/config port, 5 multiplexed ) ? The board has no R...
...against bruit force and other attacks highly stable and reliable below specs: # VoIP calls with auto QoS using the SIP protocol standards (both incoming and outgoing calls) # Android OS: - and above # Connection: GPRS, EDGE, WiFi, 3G, 4G (all above 15 kbits) # Built-in tunneling and encryption (for both the signaling and the media) # Codec: G.711 (PCMU and PCMA), GSM, Speex, iLBC, G.729 G.723 # 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 # System phone-book # Call history # Balance and rating display, call timer, status logs, detailed logs # Multi-tasking supp...
...against bruit force and other attacks highly stable and reliable below specs: # VoIP calls with auto QoS using the SIP protocol standards (both incoming and outgoing calls) # Android OS: - and above # Connection: GPRS, EDGE, WiFi, 3G, 4G (all above 15 kbits) # Built-in tunneling and encryption (for both the signaling and the media) # Codec: G.711 (PCMU and PCMA), GSM, Speex, iLBC, G.729 G.723 # 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 # System phone-book # Call history # Balance and rating display, call timer, status logs, detailed logs # Multi-tasking supp...
I want someone to code using MIKROC for dsPIC . details : PIC waits for a sting from serial port, based on the string the PIC would look for a .wav file and play it from the SD Card. I would need the coding to be done in MikroC for dspic.
...against bruit force and other attacks highly stable and reliable below specs: # VoIP calls with auto QoS using the SIP protocol standards (both incoming and outgoing calls) # Android OS: - and above # Connection: GPRS, EDGE, WiFi, 3G, 4G (all above 15 kbits) # Built-in tunneling and encryption (for both the signaling and the media) # Codec: G.711 (PCMU and PCMA), GSM, Speex, iLBC, G.729 G.723 # 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 # System phone-book # Call history # Balance and rating display, call timer, status logs, detailed logs # Multi-tasking supp...
We need a lightweight standard based VoIP web phone. The phone must ...Web page. The technology to implement web phone must be Java with the JavaScript API allowing us to control applet externally from the web page. Tecnical requirements: SIP and RTP stack compatible with any standard VoIP server or device like Cisco, Voipswitch, Asterix & FreeSWITCH Standard java applet Transport protocols: UDP, TCP, TLS, TCP tunnel Codec: PCMU, PCMA, G.729 & GSM (G.723 & ILBC will be valued in the selection) Possibility of Unlimited lines Click to call Server side integration using PHP Integration with any webpage & browser JavaScript API Customizable user interface, skins and languages Flexibility (all parameters/behavior must be changed/controlled by applet parame...
Hello, I Need to route a 4 layer , 4 channel bldc moto board with a DSPIC 33E. I give the schematic, but I put only 1 channel motor output, you need to duplicate the 3 others if posible. Need also to be carefule on the power. I use altium. Can you give me price ? after acceptance I can send you the full project. compoment are only on one side, and I gave you the basic placement but your are free to change. P1_drv1 connectore is not populated i am waiting for the motor pinout. Best Laurent
...Address Book Integration with local Phone Address Book AND the platform 4. User Credentials Save / Change Username & Password 5. Volume Control 6. Protocol SIP (RFC 3261) 7. Media Support RTP (Real Time Transport Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone Multifrequency) Generation & Detection (RFC 2833) 13. Call Status Indicator Displayed on phone screen 14. Call History, Call Detail Records, Current Account Balance, Current Account Balance,...
Description: Provide a VoIP Android library as a software package for embedding into an application Code-name: VoIP Android library (engine) Target platform: Android (1.5 or above) Delivery form: pre-compiled library or sources Requirements: • Supports SIP RFC standards • RTP/RTPC(UDP), an effective jitter buffer implementation • G.711 (A-Law and u-Law), iLBC and optional G.729A/B • Echo Cancellation support, Tone Generator, File playback • Call recording (optional) • STUN support for NAT/firewall traversal (optional) • Supports multiple SIP profiles (optional) • Simultaneous outbound calls and conferences supporting • Ability to configure the cellular ...
Making and receiving SIP calls Video calling using H264 and H263+ codecs Multitasking, receiving calls in background and sleep mode (using TCP connection) Multitasking, receiving calls in background mode using Keep Awake feature Integrated contacts from the phone Balance information Destination, rate information Time, call status logs Last ca...Multitasking, receiving calls in background and sleep mode (using TCP connection) Multitasking, receiving calls in background mode using Keep Awake feature Integrated contacts from the phone Balance information Destination, rate information Time, call status logs Last calls log Redial Hold Sending and receiving SMS Echo cancellation Supported codecs: Audio G729, g722, G711, Speex (16 KHz), iLBC and GSM. Sign up link Rec...
Making and receiving SIP calls Multitasking, receiving calls in background and sleep mode (using TCP connection) Multitasking, receiving calls in background mode using Keep Awake feature Integrated contacts from the phone Balance information Destination, rate information Time, call status logs Last calls log Redial Hold Sending and receiving ...Multitasking, receiving calls in background and sleep mode (using TCP connection) Multitasking, receiving calls in background mode using Keep Awake feature Integrated contacts from the phone Balance information Destination, rate information Time, call status logs Last calls log Redial Hold Sending and receiving SMS Echo cancellation Supported codecs: Audio G729, g722, G711, Speex (16 KHz), iLBC and GSM. Sign up link Rec...
...party user databases via HTTP 3. Address Book Integration with Phone Address Book 4. User Credentials Save / Change Username & Password 5. Volume Control 6. Protocol SIP (RFC 3261) 7. Media Support RTP (Real Time Transport Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. Address Translation (NAT) Traversal Through Encryption & STUN Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone Multifrequency) Generation & Detection (RFC 2833) 13. Call Status Indicator Displayed on phone screen 14. Call History ,Call Detail Records,Current Account Balance,Current Account Balance,Current Account Minutes Remain...
...databases via HTTP 3. Address Book Integration with Phone Address Book 4. User Credentials Save / Change Username & Password 5. Volume Control 6. Protocol SIP (RFC 3261) 7. Media Support RTP (Real Time Transport Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. Address Translation (NAT) Traversal Through Encryption & STUN Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone Multifrequency) Generation & Detection (RFC 2833) 13. Call Status Indicator Displayed on phone screen 14. Call History ,Call Detail Records,Current Account Balance,Current Account Balance,Current Account Minutes Rem...
Hi, I need help implementing a cross-correllator and autocorrelator across two channels. I currently have a dsPIC Starter Kit with the 33FJ256GP506 which I would like to base the code around. I'm eventually going to have a custom board built up with the uC and associated periphials. If you have experience with the cross-correlation function, I would be interested in outsouring the firmware portion of the project.
...touch (32GB and 64GB models from late 2009) •Call display, destination per minute cost display (once send clicked) , real time cost display (when call connected) •Speakerphone, Mute and Hold •Call history – list of received, missed and dialed calls (with call cost) •Call recording •Contact List and Contact Favorites – leveraging the iPhone Contacts •Audio codecs include G.711, G.722, G.729, iLBC, GSM and SILK •Support for DTMF: the ability to enter numbers to use an auto attendant Via RFC 2833, SIP INFO and Inband •VPN Support •Language: English •Network traversal o Application Managed, Server Managed or User Specified o Global IP Support o STUN & ICE • Media Efficiency and Quality o Noise Re...
...touch (32GB and 64GB models from late 2009) •Call display, destination per minute cost display (once send clicked) , real time cost display (when call connected) •Speakerphone, Mute and Hold •Call history – list of received, missed and dialed calls (with call cost) •Call recording •Contact List and Contact Favorites – leveraging the iPhone Contacts •Audio codecs include G.711, G.722, G.729, iLBC, GSM and SILK •Support for DTMF: the ability to enter numbers to use an auto attendant Via RFC 2833, SIP INFO and Inband •VPN Support •Language: English •Network traversal o Application Managed, Server Managed or User Specified o Global IP Support o STUN & ICE • Media Efficiency and Quality o Noise R...
I need a custom firmware for Vonets V600 - The firmware should support SIP 2.0, TCP/UDP/IP, RTP/RTCP, HTTP, DNS, DHCP, NTP, TFTP Codecs: G.723.1, G.729A/B, G.711 (a-Law and u-Law), G.726, G.728, iLBC Support standard voice features such as Caller ID Display or Block, FLASH, in-band and out-of-band DTMF (RFC2833), Dial Plans, off-hook auto dial. And additional features that will discussed in future and for additional cost.
...ITSP Softswitch Solution for callback with the following attributes; General Handling at least over 10000 Concurrent Calls at a given point in time Service setup takes just few hours Real-Time Traffic Control and Rerouting Full Customer Support Billing and Monitoring by Customers Full Support of H.323 and SIP Full Support of g711, g729, gsm, speex, ilbc and other codecs Media Transcoding (Codec Conversion) H323 to SIP (and vice-versa) Conversion Billing & Accounting Prepaid and Postpaid Customer Management Real Time Balance Update Balance Dependent Routing ASR/ACD Dependent Routing Settings Customer Credit Limits Payments and Billed Minutes Consolidated Report Automatic In...
## Job Description i m looking for an iphone expert who could take any open source implementation of a sip softphone for iphone/ipad and rebrand the UI a little bit. Following codecs shall be supported G.729 , G.722, g729,g711u/a, ilbc. probably you like to start from one of the following, if not something better then these as you are the expert. [][1]/ [][2] All source code is required and need to be managed in our svn plus i need build able instructions. if you are interested please share your last iphone,cocoa,objective C work and i am open to here what it will cost to deliver the solution, the max budget i have put is only an example, the fast you deliver the better is your
...capabilities to web applications with a Javascript API . Something like The idea is to host the "core" jar files in our media servers: our customers can open a popup (link) in their websites that will call our domain () in order to use the webphone. Fully skinable (http/css) HDvoice VPN support - Support multiple voice codecs (G711 ,G723 ,G729, GSM,speex, iLBC) for sending audio stream. - DTMF: Inband / Outband / SIP INFO - Auto Answer - CLI Support - Possibility to run multiple instances on the same machine - Work with all standard SIP servers, mac,os x linux windows and all broweser compatable SIP Parameters can be encripted and inserted to the java applet from an HTML file passed on asparameters to it (user, passWord, sipServer) that way the
...capabilities to web applications with a Javascript API . Something like The idea is to host the "core" jar files in our media servers: our customers can open a popup (link) in their websites that will call our domain () in order to use the webphone. Fully skinable (http/css) HDvoice VPN support - Support multiple voice codecs (G711 ,G723 ,G729, GSM,speex, iLBC) for sending audio stream. - DTMF: Inband / Outband / SIP INFO - Auto Answer - CLI Support - Possibility to run multiple instances on the same machine - Work with all standard SIP servers, mac,os x linux windows and all broweser compatable SIP Parameters can be encripted and inserted to the java applet from an HTML file passed on asparameters to it (user, passWord, sipServer) that way the
...Pc-2-phone • Voicemail • API with Paypal. • All service need to communicate with my Relay server to by pass NAT. Gateway to Interconnect with Carriers: H323 / SIP / Codec support G.711uLaw, G.729, G7231, iLBC, GSM, G726, H264, and H263 for both Voice & Video Applications. IPSec Tunnel Live Active calls monitoring / Call Logs / Failed calls / Presence - to see who is online / enables rapid diagnosing and resolution of issues with the added compatibility of Wireshark and Ethereal. Call trace for support, debug, and diagnostic. Admin & User web Portal Interface. Audio codecs: iLBC, Enhanced G.711, iSAC, iPCM-wb, andover 20 other standard codecs Video codecs: LSVX, H.263, MPEG-4, H.264, H.264 SVC(scalable video codec) tool allo...
- Fully featured SIP client - Support multiple voice codecs (G711 ,G723 ,G729, GSM, iLBC) for sending audio stream. - DTMF: Inband / Outband / SIP INFO - Auto Answer - CLI Support - Possibility to run multiple instances on the same machine - Work with all standard SIP servers Android application
Dear Freelancer, We need fully functional SIP/jingle client completes with the UI for Desktop Windows XP/Vista/7. A...SIP/jingle client completes with the UI for Desktop Windows XP/Vista/7. All features required: 1. Skinnable UI. 2. Interoperable with FreeSWITCH. 3. Supports login, presence and status, contacts, interaction with web service, text chat, voice chat, voice conference and video chat functionalities. 4. Supports NAT traversal such as STUN and ICE. 5. Supports Voice codec such as G711, GSM, Speex, iLBC and G729. 6. Programming languange C/C++ or .NET. The project will be evaluated based on: 1. Quality. 2. Support. 3. Price. 4. Time. The delivery of the project should include source code. The project will have support for bugs fixing. Thank you for ...
To build the hardware up and program the chip. Testing will be provided by user. -Hardware schematic and software codes will be provided. Will require the user to understand about H-bridge operation and Program dspic chip using MPlab IDE. Note: Only require those that are able to finish the Project Fast and able to finish it within 2 - 3 days.
...background support for iOS4 o Transfer and attended transfer o Push Notifications, a reliable way to receive calls when the Softphone is closed o Multi line o Call waiting o Call Grouping o Conference calling o Customizable ringtones, choose from our selection or pick something from your media library o HD Wideband audio through G.722 o excellent sound quality, includes the G.711, GSM and iLBC audio codecs. Make an in app purchase to add G.729 Annex A for great quality over 3G networks o Completely redesigned audio optimized for the best VoIP experience on the iPhone o Number rewriting, enabling you to utilize your existing contacts without having to create new entries to satisfy the dialing requirements of your PBX or SIP provider o Address Book Matching to automatic...
Need a BlackBerry SIP based VoIP Dialer. Obviously full duplex is a must (which is a known challenge on Blackberry). Core Features: - Connection over UDP - Standard Authentication using MD5 challenge - Standard Codecs (Speex 8/16/32), ILBC, GSM, PCMU, PCMA - Echo Cancellation I'm open to using open source SIP Libraries if needed, but please give me a time/cost estimate using both open source and non open source SIP Libraries. Thanks
Description: Provide a VoIP Android library as a software package for embedding into an application Code-name: VoIP Android library (engine) Target platform: Android (1.5 or above) Delivery form: pre-compiled library or sources Requirements: • Supports SIP RFC standards • RTP/RTPC(UDP), an effective jitter buffer implementation • G.711 (A-Law and u-Law), iLBC and optional G.729A/B • Echo Cancellation support, Tone Generator, File playback • Call recording (optional) • STUN support for NAT/firewall traversal (optional) • Supports multiple SIP profiles (optional) • Simultaneous outbound calls and conferences supporting • Ability to configure the cellular ...
... 2. Ethernet Ports: 1 x WAN, 1 x LAN (10/100Mbps, integrated PoE). PSTN Life Line Ports: 2 PSTN fail-over life lines. Peripheral Ports: USB, Audio In, Audio Out. Conference Rooms: 2. Unified Message Storage: 75 hours of voicemail, 5000 fax pages, 2 hours of video mail. Registered Extensions: 100. Voice Codecs: G.711, G.723, G.729 A/B/E, G.726, iLBC, T.38 fax relay. Video Codecs: H.264, H.263/H.263+. Communication/Security Protocols: TCP/UDP/IP, RTP/RTCP, ICMP, ARP/RARP, DNS, DDNS, DHCP, NTP, TFTP, TELNET, HTTP/HTTPS, PPoE, SIP(RFC3261),STUN, SRTP, TLS/SIP Compliance: FCC: Part 68 & 15B. CE: EN55022, EN55024, TBR21, EN60950, C-Tick: AS/NZX CISPR22, CIS PR24 ...
...handsets Symbian S60 3rd edition SIP Softphone Features: 1. Balance Display, call timer 2. Phone Book 3. Recent Calls (dialed and revived, missed calls) 4. PC to PC and pc to phone 5. User Management (save user information in encrypted form in flat file) 6. Hardcoded SIP server 7. Skin Support 8. Vista, windows 7 support 9. Codec support with selection (gsm, g711, g729, speex, ilbc) 10. DTMF support 11. Audio device selection wizard 12. In coming call alert 13. DND, Hold, Auto Answer 14. Call Transfer and forwarding...
