Softphones srtp jobs
...voice and SMS with instant messaging or chat, live video sharing and file transfer across all devices and networks Standalone Messaging 1-to-1 Chat Group Chat File Transfer Content Sharing Social Presence Information IP Voice call Best Effort Video call Geolocation Exchange Network based blacklist Capability Exchange based on Presence or SIP OPTIONS UDP, TLS, TCP, STUN, TURN, ICE, RTP, SRTP, Relay proxy Phase 1 : Server + iOS + Android = 3 weeks delivery Phase 2 : PC + Mac = 2 weeks delivery Phase 3 : Windows Phone + Kindle = 2 weeks delivery Phase 4 : BB + Nokia = 2 weeks delivery Payment will be subject to delivery of live app after each phase with satisfactory and completion test for all features inclusively. NO upfront or partial payment will be made. ...
I own a Synology server on which I installed the asterisk package. It works correctly, but I have to add a sip voip service that requires coding and SRTP tls but are not currently installed. I am looking for someone to install them.
...TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets Conferencing for up to 8 parties Peer-to-peer media transport over UDP-enabled networks for low latency and high scalability Scalable to tens of millions of subscribers Security: TLS for signalling encryption SRTP for media encryption THANK YOU...
...TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets Conferencing for up to 8 parties Peer-to-peer media transport over UDP-enabled networks for low latency and high scalability Scalable to tens of millions of subscribers Security: TLS for signalling encryption SRTP for media encryption THANK YOU...
...TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets Conferencing for up to 8 parties Peer-to-peer media transport over UDP-enabled networks for low latency and high scalability Scalable to tens of millions of subscribers Security: TLS for signalling encryption SRTP for media encryption THANK YOU...
...TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets Conferencing for up to 8 parties Peer-to-peer media transport over UDP-enabled networks for low latency and high scalability Scalable to tens of millions of subscribers Security: TLS for signalling encryption SRTP for media encryption YOU...
Telemarketing to source for prospective clients. Do you think you speak well? Are you able to converse easily with strangers? Do you possess basic microsoft excel skills? Do you have experience working with any softphones (ie. zoipper)? If you do, this job suits you. I am looking for a short term telemarketer right now. If you think you have what it takes, look no further!
Have a look at my project regarding VOIP. Freeswitch 1.2.23 Configured as Media Server - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones () - Voicemail to Email - SIP presence configured (alternatively configured on Kamailio) - SIP Instant Messaging (alternatively configured on Kamailio) Kamailio 4.1 Configured as SIP Frontend Server in front of Freeswitch - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones (http://www
Have a look at my project regarding VOIP. Freeswitch 1.2.23 Configured as Media Server - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones () - Voicemail to Email - SIP presence configured (alternatively configured on Kamailio) - SIP Instant Messaging (alternatively configured on Kamailio) Kamailio 4.1 Configured as SIP Frontend Server in front of Freeswitch - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones (http://www
Please find requirement below: Freeswitch 1.2.23 Configured as Media Server - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones () - Voicemail to Email - SIP presence configured (alternatively configured on Kamailio) - SIP Instant Messaging (alternatively configured on Kamailio) Kamailio 4.1 Configured as SIP Frontend Server in front of Freeswitch - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones
...create must provide communications in three styles: - Regular SIP based telephone calls - SIP based PTT mode (sender and receiver in application) - Short message system (based on external XMPP server) Basic application requirements are: - Android Jelly Bean, custom cell phone design, screen size 4,3” with dedicated PTT button - VoIP Calls and PTT must complain SIP standard - VoIP Calls use SRTP - Addidtional directory Services from VoIP Server. Server not part of this work, interface descriptions are given - All communication will be over LTE or WIFI, adaptive codecs must be available - Adaptive GUI design, colored scheme, intuitive usage. Screen design must be included. - All source code work shall be delivered in well documented form. The complete Server S...
We want the users/clients on A2billing-Asterisk server to be able to receive calls on their softphones and their IP phones. Presently all client with SIP created for are able to make calls using the softphones and the IP phones but non is able to receive income calls either between each other or from inbound DIDs. So we need some one that can edit the configuration to be able to do the above. This is a production server that is life 24/7. Thanks,
...TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets Conferencing for up to 8 parties Peer-to-peer media transport over UDP-enabled networks for low latency and high scalability Scalable to tens of millions of subscribers Security: TLS for signalling encryption SRTP for media encryption...
I have Asterisk 11.8.1 running with FreePBX 2.0.11.32. I have the new WebRTC module installed but I am having issue with SRTP Packets. I receive the following errors when making calls: [2014-04-09 08:43:29] WARNING[18338][C-00000014] sip/sdp_crypto.c: Could not set SRTP policies [2014-04-09 08:43:29] WARNING[18338][C-00000014] sip/sdp_crypto.c: Could not set SRTP policies [2014-04-09 08:43:29] WARNING[18338][C-00000014] chan_sip.c: Rejecting secure audio stream without encryption details: audio 25176 RTP/SAVPF 111 103 104 0 8 106 105 13 126 I need this asap
Is it possible to integrate the CRM Vtiger with the PBX service Call Centric and one of the following softphones Zoiper or X-Lite. If so I would like it set up ASAP. PBX PROVIDER: DIALERS: or CALL CENTRE CRM: That is the ideal set up I'm looking for. Unless you can think of a better CRM PBX ...
...==> Cable modem ==> DDWRT router ==> my computers I would like to somehow use my connection to provide external access to three different servers located in my home. I will have the following servers: 1. Vicibox/Vicidial server (based on Asterisk) where users can use their softphones to register to. Will have its own static IP. 2. A separate Asterisk server where users can use their softphones to register to. Will have its own static IP. 3. An OpenSIPs/Asterisk server where users can use their softphones to register to OpenSIPs. Will have its own static IP. My issue is I am unsure whether I need 4 separate internet connections to make this work (one for personal, and one for each of the servers). Hopefully there is a way I can do some sor...
...NOT bid if you are not 110% confident you ALREADY KNOW how to do this. Your bid MUST INCLUDE the # of hours you expect this to take. In addition to the video, some other bullet points are: - For calls that don’t pick up after X seconds, disconnected #s, busy tones, the system should just skip over those and mark those as called. - Need to be able to set caller ID for each of the softphones registered - Multiple simultaneous users may be in sessions simultaneously (user 101 shown in the example, but could also have folders for 102, 103,...)...
I need you to create the ssl certificates and setup freepbx and aastra and yealink ip phones so that they will communicate using TLS for sip and SRTP for rtp packages. Only bid if you have experience with an identical scenario. MAX budget is $40! Please do not place higher bids if you want to be considered. Thanks
We are looking for a person with experience and knowledge in deploying VoIP solutions. 1) This project involves deploying a Multi-tenant PBX solution. 2) Kamailio will Load balance traffic between the FreeSWITCH servers located in different locations. 3) Managing users will be done through a central PostgreSQL Replicated DB 4) Users will connect via TLS to Kamailio / SRTP for the media Please feel free to contact me for further Questions. Thanks! David
...DTMF: the ability to enter numbers to use an auto attendant Via RFC 2833, SIP INFO and in-band Advanced Features Network Traversal Application Managed, Server Managed or Custom Configuration Global IP Support STUN & ICE Media Efficiency and Quality Noise Reduction Echo Cancellation Voice Activity Detection Quality of Service Codecs Variable Bitrate Security and Encryption TLS and SRTP (Secure call signaling and audio encryption) DNS SRV record lookups Call quality statistics VPN support...
We are looking for a Voip Pro that can setup a Freeswitch cluster using FusionPBX. Requirements: 1. Quick delivery - before Christmas. 2. SIP TLS + SRTP. 3. Centrally managed DB for both freeswitches 4. Presence enabled. 5. Documentation & knowledge transfer Attached file. (Please see SETUP A)
We are looking for a person with experience and knowledge in deploying VoIP solutions. The idea is to create a sip Loadbalancer that handles the sip traffic and routes the RTP to Freeswitch servers located in multiple sites. the system needs to support SRTP and hold a central DB for the directory Users and Dialplans. Please feel free to contact me for further Questions. Thanks ! David
Implementing advanced Asterisk clusters including load balancing, TLS, SRTP, Encrypting the Sip trunks between servers in Amazon Cloud environment. In later stage we will be doing Opensips implementation to back end with Asterisk requiring knowledge of dial plans, load balancing, rtp media proxy.
I need a SIP dialer application for iOS and Android with the following features. - Can use any opensource SIP SDK for developement (Linphone can be used as open source) - Support fo...application for iOS and Android with the following features. - Can use any opensource SIP SDK for developement (Linphone can be used as open source) - Support for G729, SILK, and G711u - Should work with any standard SIP Soft Switch (Asterisk) - Soft Switch's IP Address should be embedded in the Code. - User should be prompted to give in Username and Password Only. - Should support TLS/SRTP - I need the dialer to work with A2Billing. - Account Balance and Calling destination should be displayed on the screen upon dialing the number. - Should have the functionality of working in b...
SIP-based phone for Apple IOS (Android, Blackberry, Windows Mobile) that uses a Wi-Fi or cellular data network connection to make and receive calls. Similar to Counterpart Bria or Acrobits Softphone, but customized. Feature...for Apple IOS (Android, Blackberry, Windows Mobile) that uses a Wi-Fi or cellular data network connection to make and receive calls. Similar to Counterpart Bria or Acrobits Softphone, but customized. Features and support for: - standard protocols SIP(S) - Audio codecs: G.711 a/u, G.722 (HD), iLBC, GSM, OPUS and SILK - Network Traversal - Security and Encryption (AES128/AES256), TLS and SRTP (Secure call signaling and audio encryption), ZRTP - VPN support - Remote management/configuration - Encrypted messaging Compatible with: - Asterisk...
...the telemarketing team). if someone is calling that number, then it should get routed to PBX where the incoming call can get routed by pressing 101, 102, 103 to go to individual employees. ( 5 employees with Skype number 347-xxx-xxx and 5 with the Skype Connect Sip account number. Only these last employees will use Vtiger for now (only Telemarketing team (5 Employees).) ).Employees have got a softphones installed on their computers (X-lite3). We need this virtual PBX to do the following: a. 718-xxx-xxxx is our company number. if someone calls that number, then it will prompt for 101 to employee 1 , 102 to employee 2 etc. 100 for operator etc. ( Same for the Skype Connect Number (Voip number), for the Telemarketing team) b. We eventually will also add more employees. c. Some ...
Looking for someone to help guide me through a few questions with SIP phones. This could end up being a longer term relationship as we will be using VoIP more over the next couple months. My current issue today is wanting to transfer two numbers so that incoming calls to those numbers go to softphones on computers here in the office. What do we need to make this happen?
Hello! I am looking for an individual to help setup an IVR on my Asterisk PBX. It is connect using the Google Voice Modif and I already have some extension setup, but right now I can only route the inbound call to...and it doesn't start the IVR system recording that I made from a text to speech to MP3 program. I need it to say. "Welcome to CentricStores, If you know your parties extension, please enter it now, otherwise for sales press 1, for customer service press 2," and so on... I don't have a super big budget, and this is connected on my home office. I plan on getting regular ip sip phones instead of softphones shortly, but I expect to have about 10 phones on this system. I have read forums about people getting this to work, by routing the Google Voice number...
...call quality Support for DTMF: the ability to enter numbers to use an auto attendant Via RFC 2833, SIP INFO and in-band Advanced Features Network Traversal Application Managed, Server Managed or Custom Configuration Global IP Support STUN & ICE Media Efficiency and Quality Noise Reduction Echo Cancellation Voice Activity Detection Quality of Service Security and Encryption TLS and SRTP (Secure call signaling and audio encryption) DNS SRV record lookups Call quality statistics VPN support Accessories Headset with microphone Headphones (no microphone): Bluetooth™ support: dependent on Android device and operating system...
...(most produce 150 + sales per per month) payment is paid weekly by bank wire, paypal or western union we will pay as you go as you meet our agreed bench marks. We have call centers making alot of money with this ongoing project and you should make alot also. Scripts, Rebuttals, Product Info provided. rnWE will also provide you with all live incoming leads and voip softphones. It is important that all agents must speak Fluent English. rnrnView our exciting product video to understand what the system does @ rnVisit our website @ rnrnThank You. rnrnGary Parker
...are looking for call centers that can produce 30-50 sales a month, payment is paid weekly by bank wire, paypal or western union we will pay as you go as you meet our agreed bench marks. We have call centers making a lot of money with this ongoing project and you could make a lot also. Scripts, Rebuttals, Product Info provided. We will also provide you with all live incoming leads and voip softphones. It is important that all agents must speak Fluent English. Thank You Maria - Director About Us: We are an Australian company who growing day by day. Most of our dealings are online and we have new products launching every quarter. This new venture has a goal is to reach 500 listings by the end of the year. The basic listings are $250, and you will earn $30-$50...
...looking for call centers that can produce 30-50 sales a month, payment is paid weekly by bank wire, paypal or western union we will pay as you go as you meet our agreed bench marks. We have call centers making a lot of money with this ongoing project and you could make a lot also. Scripts, Rebuttals, Product Info provided. We will also provide you with all live incoming leads and voip softphones. It is important that all agents must speak Fluent English. Thank You Maria - Director About Us: We are an Australian company who growing day by day. Most of our dealings are online and we have new products launching every quarter. This new venture has a goal is to reach 500 listings by the end of the year. The basic listings are $250, and you will earn $30-$50...
The project is creating a softphone that supports: direct IP-to-IP calls, phonebook, SIP,SRTP. works on windows and android.
...(most produce 150 + sales per per month) payment is paid weekly by bank wire, paypal or western union we will pay as you go as you meet our agreed bench marks. We have call centers making alot of money with this ongoing project and you should make alot also. Scripts, Rebuttals, Product Info provided. rnWE will also provide you with all live incoming leads and voip softphones. It is important that all agents must speak Fluent English. rnrnView our exciting product video to understand what the system does @ rnVisit our website @ rnrnThank You. rnrnGary Parker
The project is creating a softphone that supports: direct IP-to-IP calls, phonebook, SIP,SRTP. works on windows and android.
Install ejabberd on Amzon server for XMPP service on Bria softphones I hav etried openfire with no luck and have been told ejabbered is the best product
I would like to have srtp installed on my synology to work with Asterisk. Asterisk is running. It is a standard Synology Nas, no compilers installed yet
I would like to have srtp installed on my synology to work with Asterisk.
Hi, we need a sip dialer for Symbian S60 series and other possibly . Major features required are as follows 1. Stun support 2. SIP with TLS ( SRTP is not needed just SIPS) 3. G729 codec Core need is for Symbian , but in addition to Symbian Andriod and IPhone version are also welcome.
Object Description: We need someone to set up voice broadcasting software on our server. We need someone to: Develop, configure software using Asterisk, Vicidial or other to send voice broadcasting. You will set up through remote access to our Host Server You will show us how to Manage and Initiate campaigns Voice...maintenance then run the raided or mirrored backup till remedy for the primary drive/drives is complete. Any demo would be very helpful if available. My uses for the system are heavily utilized for robo calling thousands of phone numbers an hour delivering our automated messages to people that answer the phone then press 1 to be transferred to a live agent. Calls get routed to our voip softphones as the telemarketers communicate with the potential cl...
Object Description: We need someone to set up voice broadcasting software on our server. We need someone to: Develop, configure software using Asterisk, Vicidial or other to send voice broadcasting. You will set up through remote access to our Host Server You will show us how to Manage and Initiate campaigns Voice...maintenance then run the raided or mirrored backup till remedy for the primary drive/drives is complete. Any demo would be very helpful if available. My uses for the system are heavily utilized for robo calling thousands of phone numbers an hour delivering our automated messages to people that answer the phone then press 1 to be transferred to a live agent. Calls get routed to our voip softphones as the telemarketers communicate with the potential cl...
Object Description: We need someone to set up voice broadcasting software on our server. We need someone to: Develop, configure software using Asterisk, Vicidial or other to send voice broadcasting. You will set up through remote access to our Host Server You will show us how to Manage and Initiate campaigns Voice...maintenance then run the raided or mirrored backup till remedy for the primary drive/drives is complete. Any demo would be very helpful if available. My uses for the system are heavily utilized for robo calling thousands of phone numbers an hour delivering our automated messages to people that answer the phone then press 1 to be transferred to a live agent. Calls get routed to our voip softphones as the telemarketers communicate with the potential cl...
...running Asterisk 1.8.10.1 and FreePBX v2.11.0.0rc1.8. The system is setup and ready for configurations. I have Cisco 7960, 7971 SIP based IP phones and a Cisco SPA303 that I would like have configured with phone extensions. I have the most of the configuration files setup for the Cisco 79xx phones but need someone to review them. If we can get away with using the Cisco SPA303 and get some VOIP softphones like 3CX to register and work that would be great. Basic features needed: Extensions, Paging/Intercom, Ring Groups, and Voicemail Parking, Music on Hold. IVR would be nice. Just need someone familiar with the GUI and where to set things up to help. Remote access can be granted to the server and I can update any DHCP TFTP configs if needed. I already have the TFTP server running ...
I need some one to help me to create a PBX server using Asterisk Now, I already have a server, the card () the network is ready, the softphones, etc
I am looking for someone with experience in both Asterisk and Microsoft Lync Server. We need to be able to make/receive external phone calls from Microsoft Lync through our Elastix PBX (Asterisk). Currently we use Softphones (XLite and Bria) to make phone calls through Asterisk and we run Microsoft Lync for internal IM/VC/Presence. We want to be able to remove the need for the Softphones and make and receive phone calls directly through Lync. We have two Elastix machines, one in each of our offices (Sydney - Australia AND Manila - Philippines). We are running one Lync server in Sydney - Australia. This project is to enable calling from Sydney based Microsoft Lync users through the Sydney based Elastix machine (unless more advanced options are presented). For security reas...
REMEMBER: THIS PROJECT IS NOT FOR ANY FRESHERS/JUNIOR DEVELOPER..SO PLEASE DON'T BID IF YOU DON'T HAVE ANY PREVIOUS EXPERIENCE IN VOIP/SOFTPHONE.. WHAT I need is already developed code with following features. Basic softphone features , 1-registration(UDP,TLS,TCP),(RTP,SRTP) (BASIC) 2-dial a number ,receive a call (BASIC) 3-the normal feature that a softphone has(speaker ,voice mail button, conference,dial from phonebook)…. This will discuss it later (optional) Basic Main Feature Our GSM normal dial plan is 03XXXXXX What I want to see is the scenario below : (BASIC) PBX send a call(invite 03045212@UA Registered IP address) to the registered UA which is your softphone in this case ----à Registered UA(Your Softphone)---àt...
OBJETIVO Evaluar la calidad de los codecs de audio y video integrados en softphones SIP para PCs. DESCRIPCIÓN Los softphones son aplicaciones de software desarrolladas para PC que permiten establecer una comunicación de audio y video. Algunos vienen con otras capacidades adicionales pero éstas no importan al objetivo propuesto. El funcionamiento básico de un softphone consiste en establecer una llamada con otro terminal utilizando señalización SIP a través de la red IP, y luego de negociar las capacidades de medios con SDP, tanto de audio como de video, comenzar a enviar y recibir flujo RTP de ambos tipos. QUÉ DEBE HACER EL INSTRUMENTO Para evaluar la calidad de los codecs de audio y video, se deber...
Actually I am looking for a VOIP application..This project is not for time serious bidders i need.. Those who have made already basic soft phone ,I will give priority to select them..don't bid if you don't have any previous experience of voip project.I will report as a spam if you place only copy paste bid.. Basic softphone features which i need is--- 1-registration(UDP,TLS,TCP),(RTP,SRTP) (BASIC) 2-dial a number ,receive a call (BASIC) 3-the normal feature that a softphone has(speaker ,voice mail button, conference,dial from phonebook)…. This will discuss it later (optional) Basic Main Feature Our GSM normal dial plan in Lebanon is 03XXXXXX (03045212 as its my GSM local number) What I want to see is the scenario below : (BASIC) PBX send a call(invite...
...Actually I am looking for a VOIP application..This project is not for time serious bidders i need.. Those who have made already basic soft phone ,I will give priority to select them..don't bid if you don't have any previous experience of voip project.I will report as a spam if you place only copy paste bid.. Basic softphone features which i need is--- 1-registration(UDP,TLS,TCP),(RTP,SRTP) (BASIC) 2-dial a number ,receive a call (BASIC) 3-the normal feature that a softphone has(speaker ,voice mail button, conference,dial from phonebook)…. This will discuss it later (optional) Basic Main Feature Our GSM normal dial plan in Lebanon is 03XXXXXX (03045212 as its my GSM local number) What I want to see is the scenario below : (BASIC) PB...
