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  • Voice Communication over TCP/IP

    - by Micha
    Hello, I'm currently developing application using DirectSound for communication on an intranet. I've had working solution using UDP but then my boss told me he wants to use TCP/IP for some reason. I've tried to implement it in pretty much the same way as UDP, but with very little success. What I get is basically just noise. 20% of it is the recorded sound and the rest is just weird noise. My guess for the reason is that TCP needs to read all the accepted data several times until it gets the final sound I can play. Now two questions: Am I on the right tracks? Is it even good idea to use TCP/IP for this kind of application (voice conferencing of sorts)? I'm doing it in C# but I don't think this is language specific.

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  • Tips for creating a video-voice-chat application

    - by Marco
    I want to create a simple chat application that supports voice and video (something like Skype or Google Talk). I don't want to write everything from scratch, so my question is do you know some good libraries for that? I stumbled over libjingle (c++) and Smash (Java), both implementing the XMPP extension Jingle. Would you recommend one of those?

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  • Need direction in creating a voice chat application

    - by WarDoGG
    I want to create a voice chat application as a part of a project. However, i am totally lacking direction regarding the programming language to use, the technologies involved. Can somebody please guide me as to how i should proceed ? Here are the features that i require : user to user voice chat ability to chat in conference (more than 2 users) How do i connect one user to another ? How to handle voice transmission ? How to effectively route packets in a conference ? I'm thinking the most probably langauge to develop this in would be Flash. Any suggestions are welcome.

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  • Ways of breaking down SQL transactional/call data into reports -- 'square data'?

    - by RizwanK
    I've got a large database of call-traffic information (although the question could be answered with any generic data set.) For instance, a row contains : call endpoint server (endpoint_name) call endpoint status (sip_disconnect_reason) call destination (destination) call completed (duration) [duration 0 is completed] call account group (account_group) It's pretty easy to run SQL reports against the data, i.e. select count(*), endpoint_name from calls where duration0 group by endpoint_name select count(*),destination from calls where blah group by destination I've been calling this filtering or breakdown reports (I get the number of calls per carrier, etc.). Add another breakdown, and you've got two breakdowns, a la select count(*), endpoint_name, sip_disconnect_reason from calls where duration=0 group by endpoint_name, sip_disconnect_reason Of course, if you keep adding breakdowns, you end up making super-large reports and slicing your data so thin that you can't extract any trends from it. So my question is this : Is there a name for this sort of method of report writing? (I've heard words like squares, slicing and breakdown reports applied to them) --- I'm looking for a Python/Reporting toolkit that I can use to make these easier to generate for my end users. aside : Are there other ways of representing transactional data that might be useful rather than the above method? Thanks,

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  • Looking for good SIP Book

    - by Dave
    Hey guys I am looking for a SIP book similar to this one on XMPP - Professional XMPP Programming with Javascript and Jquery (http://www.amazon.com/Professional-Programming-JavaScript-jQuery-Programmer/dp/0470540710) I am new to the area and any resources would be appreciated, thanks

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  • Libraries for making a voice chat application

    - by Eric
    My development team is going to build a voice chat application. Our plan is to use a pre-made library just for this purpose, but we haven't found any good one after days of searching the internet, so I thought I would consider a question here! So the question is: What library / project do you recommend? We are deadly serious with this, so it needs to be a good working one. Preferable an open-source one as well. We have been looking at some XMPP libraries and projects, but none seems to be up-to-date, tested and well-documented.

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  • What software or service can I use to programatically make phone calls with?

    - by Jason
    I'm looking to programatically make phone call reminders to customers based upon their opt-in requests. I am NOT a telemarketer. I need to make a phone call, and play a message. I need to leave a message after the beep if an answering machine or voicemail is detected. I need to know if the message was successfully delivered. Ideally, I could offer the user feedback by pressing a button and recording their selection. I prefer Windows and .NET but would consider anything. What do you suggest?

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  • Skype arrive sur Windows Phone 8, Microsoft publie la Preview de son application de VoIP pour l'OS et une mise à jour pour iOS

    Skype arrive sur Windows Phone 8 Microsoft publie la Preview de son application de VoIP pour l'OS mobile et une mise à jour pour iOS Les utilisateurs de Windows Phone 8 peuvent déjà tester les fonctionnalités de la mise à jour de Skype pour l'OS. Microsoft vient d'annoncer la publication de la Preview de Skype pour Windows Phone 8. L'application plus épurée et moderne, est téléchargeable sur le store Windows Phone. Pour cette nouvelle mouture, Microsoft a accordé une attention particulière à une meilleure intégration avec l'interface utilisateur du système d'exploitation mobile et l'ajout de nouvelles fonctionnalités. Skype pour Windows Phone 8 apporte co...

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  • Setup LAN to serve webpages and voip and access to the web site from inside LAN with domain name

    - by Mauricio Arias
    I'd like to know if it will work: I have my domain and I´m serving a webpage in a nginx to the internet, but if I type my domain in my laptop inside LAN I access to my modem/router configuration, I cannot access to the web server unless I type the IP address. I would like to add a Bind server after the modem/router - (port forward, ports 80 and 5060), if the request is www.mydomain.com bind should resolve the nginx IP address and serve it, and if it is a voip request should address to the voip server and if I'd like to access to the website from inside LAN I'd like to type mydomain.com. Could I do it with this configuration? Do I need something else? Thanks in advace!

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  • How do i network ten branch office with voip, video calling and files sharing.

    - by Oluwalogbon
    Am an IT person, have done some networking job for my organization like Lan and wireless within the area, configure windows server to manage staff account My company has ten branch (In each state) in my country and am giving a task to connect dose branch together, which there will be VOIP, Video calling and sharing of files within the branch. I need someone to help me with this project..what and what did I need to put in place

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  • How compliant is SIP VoIP software on the net?

    - by rusbi
    I developed a SIP stack for my company. It's far from perfect, a it's lacking a lot of things from the RFCs, but it's functional and work well with a lot of tested softphones and other SIP hardware and software. My question is: How much of SIP software can truly say that they are entirely SIP compliant? (Of the softphones you can find on the internet...)

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  • How can I expire non-active sessions on my Netscreen SSG140?

    - by David Mackintosh
    I have a Juniper Netscreen SSG-140. While experimenting with a VoIP service, I defined a custom policy that was to be used to permit the possible ports in use to be sent back to the VoIP server from systems connecting across the internet. Because I'd had problems in the past with VoIP systems getting broken when their UDP sessions were expired out faster than their keep-alives were generated, I set the timeout on this custom service to be 'never'. After much experimentation, I happened to notice that my session count on the firewall has grown from a couple thousand to over 36000. After discussion with the VoIP "expert", I set the timeout to be 30 minutes; however, all the sessions set up during the experimentation process are still there, more than 3 days later. Is there a way I can force these old sessions to get expired and removed from the session table, or am I looking at resetting my firewall? (Both firewalls, actually -- they are in a cluster.)

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  • Configuring a PIX 506e for Asterisk

    - by orthogonal3
    Hi all! I'm having problems configuring a old Cisco PIX running 6.3 and wondered if anyone can lend a hand? Simply put I have a PIX 506e that I want to put in my VoIP data path. I can't update it and getting a compat version of Java for that version of PIX is tough so I can't log onto the web interface. The PIX straddles two networks..... 192.168.5.0 on the inside, ...50.0 on the outside both net masks are 255.255.255.0 I have a local Asterisk server cluster with a single service IP (<local asterisk>) SIP is on UDP 5060 and RTP (for the voip data) is on UDP 18000-18999 I know thats a big range but hey may as well. I need the 192.168.5.0 net to have web and ftp access for updates and the like. DHCP, DNS and NTP is already provided on that network so I don't need external DNS access. So I think I want the following rules: SIP or RTP from <my itsp> arriving at <outside voip ip> NATed to <local asterisk> SIP or RTP able to do the reverse route (should be covered by high sec - low sec??) HTTP and FTP access outbound for software update for the servers etc I have the following config at the minute - and I think I'm almost there (I hope)... interface ethernet0 auto interface ethernet1 auto nameif ethernet0 outside security0 nameif ethernet1 inside security100 enable password wouldyouliketobeapeppertoo encrypted passwd wouldyouliketobeapeppertoo encrypted hostname afirewall domain-name adomain fixup protocol dns maximum-length 512 fixup protocol ftp 21 fixup protocol h323 h225 1720 fixup protocol h323 ras 1718-1719 fixup protocol http 80 fixup protocol rsh 514 fixup protocol rtsp 554 fixup protocol sip 5060 fixup protocol sip udp 5060 fixup protocol skinny 2000 fixup protocol smtp 25 fixup protocol sqlnet 1521 fixup protocol tftp 69 access-list acl_ping permit icmp any any access-list voip permit ip host <my itsp> host <local asterisk> mtu outside 1500 mtu inside 1500 ip address outside <outside pix ip> 255.255.255.0 ip address inside <inside pix ip> 255.255.255.0 arp timeout 14400 global (outside) 1 <outside generic ip> nat (inside) 1 192.168.5.0 255.255.255.0 0 0 static (inside,outside) <outside voip ip> <local asterisk> netmask 255.255.255.255 0 0 static (outside,inside) <local asterisk> <outside voip ip> netmask 255.255.255.255 0 0 access-group acl_ping in interface outside access-group acl_ping in interface inside route outside 0.0.0.0 0.0.0.0 <my next hop router> 1 route outside <my itsp> 255.255.255.255 <my next hop router> 1 I think I just need a hand with the access-lists and NAT/static rules. Would anyone be able to help as I've RTFM'd the Cisco docs a few times and they're heavy. Wishing I'd completed my CCNA now! Thanks all for any help, Phil

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  • Introducción a ENUM (E.164 Number Mapping)

    - by raul.goycoolea
    E.164 Number Mapping (ENUM o Enum) se diseñó para resolver la cuestión de como se pueden encontrar servicios de internet mediante un número telefónico, es decir cómo se pueden usar los los teléfonos, que solamente tienen 12 teclas, para acceder a servicios de Internet. La parte más básica de ENUM es por tanto la convergencia de las redes del STDP y la IP; ENUM hace que pueda haber una correspondencia entre un número telefónico y un identificador de Internet. En síntesis, Enum es un conjunto de protocolos para convertir números E.164 en URIs, y viceversa, de modo que el sistema de numeración E.164 tenga una función de correspondencia con las direcciones URI en Internet. Esta función es necesaria porque un número telefónico no tiene sentido en el mundo IP, ni una dirección IP tiene sentido en las redes telefónicas. Así, mediante esta técnica, las comunicaciones cuyo destino se marque con un número E.164, puedan terminar en el identificador correcto (número E.164 si termina en el STDP, o URI si termina en redes IP). La solución técnica de mirar en una base de datos cual es el identificador de destino tiene consecuencias muy interesantes, como que la llamada se pueda terminar donde desee el abonado llamado. Esta es una de las características que ofrece ENUM : el destino concreto, el terminal o terminales de terminación, no lo decide quien inicia la llamada o envía el mensaje sino la persona que es llamada o recibe el mensaje, que ha escrito sus preferencias en una base de datos. En otras palabras, el destinatario de la llamada decide cómo quiere ser contactado, tanto si lo que se le comunica es un email, o un sms, o telefax, o una llamada de voz. Cuando alguien quiera llamarle a usted, lo que tiene que hacer el llamante es seleccionar su nombre (el del llamado) en la libreta de direcciones del terminal o marcar su número ENUM. Una aplicación informática obtendrá de una base de datos los datos de contacto y disponibilidad que usted decidió. Y el mensaje le será remitido tal como usted especificó en dicha base de datos. Esto es algo nuevo que permite que usted, como persona llamada, defina sus preferencias de terminación para cualquier tipo de contenido. Por ejemplo, usted puede querer que todos los emails le sean enviados como sms o que los mensajes de voz se le remitan como emails; las comunicaciones ya no dependen de donde esté usted o deque tipo de terminal utiliza (teléfono, pda, internet). Además, con ENUM usted puede gestionar la portabilidad de sus números fijos y móviles. ENUM emplea una técnica de búsqueda indirecta en una base de datos que tiene los registros NAPTR ("Naming Authority Pointer Resource Records" tal como lo define el RFC 2915), y que utiliza el número telefónico Enum como clave de búsqueda, para obtener qué URIs corresponden a cada número telefónico. La base de datos que almacena estos registros es del tipo DNS.Si bien en uno de sus diversos usos sirve para facilitar las llamadas de usuarios de VoIP entre redes tradicionales del STDP y redes IP, debe tenerse en cuenta que ENUM no es una función de VoIP sino que es un mecanismo de conversión entre números/identificadores. Por tanto no debe ser confundido con el uso normal de enrutar las llamadas de VoIP mediante los protocolos SIP y H.323. ENUM puede ser muy útil para aquellas organizaciones que quieran tener normalizada la manera en que las aplicaciones acceden a los datos de comunicación de cada usuario. FundamentosPara que la convergencia entre el Sistema Telefónico Disponible al Público (STDP) y la Telefonía por Internet o Voz sobre IP (VoIP) y que el desarrollo de nuevos servicios multimedia tengan menos obstáculos, es fundamental que los usuarios puedan realizar sus llamadas tal como están acostumbrados a hacerlo, marcando números. Para eso, es preciso que haya un sistema universal de correspondencia de número a direcciones IP (y viceversa) y que las diferentes redes se puedan interconectar. Hay varias fórmulas que permiten que un número telefónico sirva para establecer comunicación con múltiples servicios. Una de estas fórmulas es el Electronic Number Mapping System ENUM, normalizado por el grupo de tareas especiales de ingeniería en Internet (IETF, Internet engineering task force), del que trata este artículo, que emplea la numeración E.164, los protocolos y la infraestructura telefónica para acceder indirectamente a diferentes servicios. Por tanto, se accede a un servicio mediante un identificador numérico universal: un número telefónico tradicional. ENUM permite comunicar las direcciones del mundo IP con las del mundo telefónico, y viceversa, sin problemas. Antes de entrar en mayores profundidades, conviene dar una breve pincelada para aclarar cómo se organiza la correspondencia entre números o URI. Para ello imaginemos una llamada que se inicia desde el servicio telefónico tradicional con destino a un número Enum. En ENUM Público, el abonado o usuario Enum a quien va destinada lallamada, habrá decidido incluir en la base de datos Enum uno o varios URI o números E.164, que forman una lista con sus preferencias para terminar la llamada. Y el sistema como se explica más adelante, elegirá cual es el número o URI adecuado para dicha terminación. Por tanto como resultado de la consulta a la base dedatos Enum siempre se da una relación unívoca entre el número Enum marcado y el de terminación, conforme a los deseos de la persona llamada.Variedades de ENUMUna posible fuente de confusión cuando se trata sobre ENUM es la variedad de soluciones o sistemas que emplean este calificativo. Lo habitual es que cuando se haga una referencia a ENUM se trate de uno de los siguientes casos: ENUM Público: Es la visión original de ENUM, como base de datos pública, parecida a un directorio, donde el abonado "opta" a ser incluido en la base de datos, que está gestionada en el dominio e164.arpa, delegando a cada país la gestión de la base de datos y la numeración. También se conoce como ENUM de usuario. Carrier ENUM, o ENUM Infraestructura, o de Operador: Cuando grupos de operadores proveedores de servicios de comunicaciones electrónicas acuerdan compartir la información de los abonados por medio de ENUM mediante acuerdos privados. En este caso son los operadores quienes controlan la información del abonado en vez de hacerlo (optar) los propios abonados. Carrier ENUM o ENUM de Operador también se conoce como Infrastructure ENUM o ENUM Infraestructura, y está siendo normalizado por IETF para la interconexión de VoIP (mediante acuerdos de peering). Como se explicará en la correspondiente sección, también se puede utilizar para la portabilidad o conservación de número. ENUM Privado: Un operador de telefonía o de VoIP, o un ISP, o un gran usuario, puede utilizar las técnicas de ENUM en sus redes y en las de sus clientes sin emplear DNS públicos, con DNS privados o internos. Resulta fácil imaginar como puede utilizarse esta técnica para que compañías multinacionales, o bancos, o agencias de viajes, tengan planes de numeración muy coherentes y eficaces. Cómo funciona ENUMPara conocer cómo funciona Enum, le remitimos a la página correspondiente a ENUM Público, puesto que esa variedad de Enum es la típica, la que dió lugar a todos los procedimientos y normas de IETF .Más detalles sobre: @page { margin: 0.79in } P { margin-bottom: 0.08in } H4 { margin-bottom: 0.08in } H4.ctl { font-family: "Lohit Hindi" } A:link { so-language: zxx } -- ENUM Público. En esta página se explica con cierto detalle como funciona Enum Carrier ENUM o ENUM de Operador ENUM Privado Normas técnicas: RFC 2915: NAPTR RR. The Naming Authority Pointer (NAPTR) DNS Resource Record RFC 3761: ENUM Protocol. The E.164 to Uniform Resource Identifiers (URI) Dynamic Delegation Discovery System (DDDS) Application (ENUM). (obsoletes RFC 2916). RFC 3762: Usage of H323 addresses in ENUM Protocol RFC 3764: Usage of SIP addresses in ENUM Protocol RFC 3824: Using E.164 numbers with SIP RFC 4769: IANA Registration for an Enumservice Containing Public Switched Telephone Network (PSTN) Signaling Information RFC 3026: Berlin Liaison Statement RFC 3953: Telephone Number Mapping (ENUM) Service Registration for Presence Services RFC 2870: Root Name Server Operational Requirements RFC 3482: Number Portability in the Global Switched Telephone Network (GSTN): An Overview RFC 2168: Resolution of Uniform Resource Identifiers using the Domain Name System Organizaciones relacionadas con ENUM RIPE - Adimistrador del nivel 0 de ENUM e164.arpa. ITU-T TSB - Unión Internacional de Telecomunicaciones ETSI - European Telecommunications Standards Institute VisionNG - Administrador del rango ENUM 878-10 IETF ENUM Chapter

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  • Load balance incoming traffic

    - by justin
    Dear All Please I have the following scenario. 3 servers voip / mail / terminal one load balancing router 2 internet connections (static ip`s) My concern is to load balance incoming traffic since the outgoing traffic is being taking care by the load balancing router. For instance all offices connect to the mail server via the internet same for voip and terminal services. The mail and voip clients are set up with one of the static ip`s and the router forwards the request to the appropriate server. But obviously like this there is no fail over nor load balancing cause all requests are being directed to one internet connection. Anyone has a suggestion was thing of a dns server, does this make sens ? or maybe a hosted option ? Thanks Justin

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  • cant make outbound calls - asterisk

    - by deanvz
    I have a basic Atcom IP01 with the following config Registered Voip (SIP) Trunk Registered Voip Phone - ext Dial Plan Outbound Call rule I made use of this manual that the manufacturer supplies: http://www.atcom.cn/cn/download/pbx/ip01/ATCOM%20IP01-User%20Manual-V1.0-EN.pdf Whenever I try and make a call, it seems that the outbound call rule that i defined does not get regarded as the default rule even though the dial plan lists this as the only outbound call rule. When dialling I see in the log file the following [Jan 1 09:10:07] NOTICE[176]: chan_sip.c:14377 handle_request_invite: Call from '6001' to extension '00765243679' rejected because extension not found. The 00765243679 is a cellular number. Am I missing a configuration in order to make outbound calls? Land line, other Voip numbers and cellular calls have been tried

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  • How to choose an open source, Asterisk friendly firewall?

    - by Lucas
    I'm in pain. We are moving to a SIP based VOIP system and for whatever reason, we could not get our hosted Asterisk solution to work with our Sonicwall. Our VOIP provider gave up and is recommending an open source vendor, pfSense. A little background: We have about 30 users in our network. We use a few IPSec VPN connections for remote networks. I would like, but don't need, application layer filtering. We're active internet users, so properly traffic shaping is probably a concern. How can I tell if an open source firewall will handle VOIP setup smoothly with a hosted Asterisk system?

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  • local RTP port unreachable when using mjsip/jmf

    - by brian_d
    Hello, I create a sip session with mjsip to an external voip provider. Then I transmit a test wav file over rtp to the provider using RtpManager. The program runs with no errors and I answer the sip call. However, no audio is transmitted. When I diagnose the network traffic with wireshark, I see a bunch of RTP traffic from my localhost (behind some kind of nat) to the voip provider and nothing back. After a while I get the ICMP error "Destination unreachable (Port unreachable)" from the provider to my localhost. The software linphone works using the same localhost and voip provider - though it is using a different sip stack. Any suggestions? Thanks

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