Taller dispositivos Voip

Javier Peña Fabio Diaz John Fredy Perez

Implementación de Sistemas de Telefonía IP

Snom 760 Teléfono IP

• • • • • •

Audio de banda ancha G.711 A-law, μ-law G.722, G.726, G.729AB GSM 6.10 (full rate) Generador de ruido de confort (CNG) Detección de actividad de voz (VAD)

SIP • Cumple con RFC3261 • UDP, TCP y TLS • Autenticación codificada • PRACK (RFC3262) • Enrutamiento abierto y estricto • Indicación de código de error • Fiabilidad de las respuestas provisionales(RFC3262) • Admite la función ―early media‖ • DNS SRV (RFC3263), soporte de servidorredundante. • Oferta/respuesta (RFC3264) • Indicador de mensaje en espera (RFC3842),Suscripción a eventos MWI (RFC3265) • Control del estado de dialogo (RFC 4235) • DTMF • Cliente STUN (NAT transversal) • Admite suscripción a lista de eventos (RFC 4662) • Identificación del número llamante en más deun terminal (BLA) • Autoaprovisionamiento con PnP • Característica de presencia/lista de amigos • Admite la función ―busylampfield‖ (BLF) (Luces para indicación de ―ocupado‖) • Publicación de presencia

INFORMACIÓN GENERAL • Peso: Aprox. 920g (690g sin pie) • Certificados: FCC Clase B, Marcado CE • Seguridad: IEC 60950-1:2007, • Power over Ethernet/PoE: IEEE 802.3af, Clase 2 • Alimentación: 5 V DC (no incluido) • Consumo eléctrico: 4–7 Watt • Ethernet: Conmutador Gigabit 2 x IEEE 802.3/1 • 1 x LAN, 1 x PC: RJ 45 • Terminal: Conectores RJ-4P4C • Auricular: Conector RJ-4P4C o por USB • Puertos USB: 2 x tipo A, Interfaz de hosta velocidad alta/máxima/baja • Color: Antracita SEGURIDAD, CALIDAD DEL SERVICIO • • • • • Servidor/Cliente HTTPS TransportLayer Security (TLS) SRTP (RFC3711), SIPS, RTCP VLAN (IEEE 802.1X) LLDP-MED

Tomado de: http://downloads.snom.net/documen tation/data_snom760_es.pdf en 4 febrero 2012

CÓDECS, Audio

Taller dispositivos Voip

Características de audio • La tecnología Polycom HD Voice ofrece calidad devoz realista en cada ruta de audio - auricular,parlante manos libres y audífono1 • Parlante manos libres full dúplex Full dúplex: Tipo 1 normalizado IEEE 1329 full dúplex • Respuesta de frecuencia - 150Hz 7kHz para modosde auricular, audífono1 y parlante manos libres • Codecs: G.722 (banda ancha), G.711 μ/A, y G.729ª (Anexo B) • Control individual de volumen con info visual paracada ruta de audio • Detección de actividad de voz • Comfortnoisefill • Generación de tono DTMF / evento DTMF RTP payload • Transmisión de paquetes de audio con bajo retraso • Jitter buffers adaptables • Ocultación de pérdida de paquetes • Cancelación de eco acústico • Supresión de ruidos de fondo Red y aprovisionamiento • Switch puerto dual Gigabit Ethernet: - Puertos 10/100/1000Base-Tx a través de LAN y PC - Conforme con IEEE802.3-2005 (Claus 40) paraAnexo medios físicos - Conforme con IEEE802.3-2002 (Claus 28) paraAuto negociación enlace socio • Config de red manual o dynamic host configurationprotocol (DHCP) • Sincronización de fecha y hora con SNTP • Aprovisionamiento central FTP/TFTP/HTTP/HTTPS4basado en servidor para implementacionesmasivas.
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Soporte aprovisionamiento redundancia servidor. • Portalweb para configuración de unidad individual • Soporte QoS -- IEEE 802.1p/Q tagging (VLAN), TOSCapa 3 y DSCP • Soporte Network AddressTranslation (NAT) - estático • Soporte RTCP (RFC 1889) • Registro (log) de eventos • Mapa dígito local • Diagnósticos de hardware • Estatus y estadísticas Seguridad2 • Seguridad en la capa de transporte (TLS) • Archivos de configuración encriptados3 • Digest authentication3 • Login con contraseña • Soporte para sintaxis de URL con contraseña paraboot server4 • Aprovisionamiento HTTPS seguro4 • Soporte para ejecutables de software firmados. Alimentación • Powerover Ethernet integrada, autosensing IEEE 802.3a • Adaptador externo universal de AC (incluido 48V DC) Tomado de: http://supportdocs.polycom.com/Poly comService/support/global/pw_item_ show_doc/soundpoint_ip_670_datash eet_es.pdf en 4 febrero 2012

Avaya 1120 E DeskphoneIP

Audio Quality of Service • G.711 a-law, G.711 μ-law, G.729a and Annex B, G.722 (SIP software) • 802.1p/Q, DiffServ and VLAN tagging of telephony LAN porttraffic •Supports echo cancellation and silencesuppression Protocols • E.164 dialing • SIP Protocols: – RFC2327 — SDP: SessionDescriptionProtocol – RFC2617 — HTTP Authentication: Basic and Digest Access Authentication – RFC2976 — The SIP INFO Method– RFC3108 — Conventionsforthe use of SessionDescriptionProtocol: ATM BearerConnections – RFC3204 — MIME Media Typesfor ISUP and QSIG Objects – RFC3261 — SessionInitiationProtocol (SIP) – RFC3262 — Reliability of Provisional Responses in theSessionInitiationProtocol – RFC3263 — SessionInitiationProtocol (SIP): Locating SIP Servers – RFC3264 — AnOffer/AnswerModelwithSessionIniti ationProtocol (SIP) – RFC3265 — SessionInitiationProtocol: SpecificEventNotification

– RFC3311 — SessionInitiationProtocol (SIP) UPDATE Method – RFC3313 — PrivateSessionInitiationProtocol (SIP) Extensionsfor Media Authorization. – RFC3323 — A PrivacyMechanismfortheSessionInitiat ionProtocol – RFC3325 — PrivateExtensionstotheSessionInitiatio nProtocolforAssertedIdentitywithinTru sted Networks – RFC3329 — Security MechanismAgreement Administration and Security • Static and Full Dynamic Host Control Protocol (Full DHCP factory default) • 802.1x and Extensible AuthenticationProtocol (EAP) fordeviceauthentication and networkaccess control • Securesignalingusingstandardsbased DTLS12 • 802.1ab Link LayerDiscoveryProtocolfornetwork auto-discovery and inventorymanagement • Media pathencryptionwith RFC 3711 compliantSecure Real-time Protocol (sRTP) pre-sharedkey and publickey infrastructure13 Tomado de: http://www.avaya.com/usa/resource /assets/factsheet/1120e%20ip%20de skphone%20final%20pdf.pdf en 4 febrero 2012

Servidor proxy SIP

Brekeke SIP Server - SIP Proxy, Registrar Server Based on the Session Initiation Protocol (SIP), the Brekeke SIP Server provides reliable and scalable SIP communication platform for Enterprises and Service Providers. As it defined in the SIP baseline specification RFC 3261, Brekeke SIP Server provides functionality of SIP Registrar Server, SIP Redirect Server, and SIP Proxy Server. Brekeke SIP Server is a Stateful Proxy that maintain session status therefore performs optimum processing for call control.

SpecificationsforBrekeke SIP Server Specifications ProtocolforSignali SIP (RFC3261 ng Standard) Protocol for Delivery of Media RTP, RTCP (When using NAT Traversal)

TransportProtocol TCP, UDP RoutingMethods Registerdatabase, Dial Plan, B2BUA Proprietarymethod , UPnP

NAT Traversal Maximum Number of Concurrent Sessions

Unlimited

Administration

WebbasedAdministrati on

System Requirements for Brekeke SIP Server SystemRequirements OS Microsoft Windows, Linux, Mac OSX, Solaris Sun Microsystems JDK 1.6 or later (32bit)

Benefits - Brekeke's SIP proxy server Brekeke SIP Server provides sophisticated features that can meet requirements for creating stable and reliable SIP platform.

Java

Memory 256 MB Minimum

Registrar Service Brekeke SIP Server receives REGISTER requests from SIP UAs and updates its database appropriately. Using the registrar function, you can receive calls from any SIP UA using

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your unique SIP-URI.

Call Routing Brekeke SIP Server routes SIP requests from a SIP UA or other server, to the most appropriate SIPURI address based on its registrar database and Dial Plan. Brekeke SIP Server supports SIP redirect feature which allow servers to redirect a request back to SIP UA.

NAT Traversal Brekeke SIP Server enables SIP UAs behind the NAT to talk with other SIP UAs, including video over NAT traversal. Using NAT-enabled firewalls ensures the security that you want to retain, while giving users the ability to make media calls over the Internet between different networks. NAT TraversalFeature

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Dial Plan With a Dial Plan you can use regular expressions to define matching or filtering rules for headers and IP addresses in the SIP packets. Brekeke SIP Server’s Dial Plan increases compatibility between SIP compliant products and provide added capability for creating complex call routing.

UPnP Brekeke SIP Server can detect UPnP enabled router and control it remotely for NAT traversal. UPnPSupport

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TCP Transport Support Brekeke SIP Server added a new Transport Protocol, TCP, from version 2.2. With this support, Brekeke SIP Server is compatible with a variety of TCP enabled UAs, servers, and applications. TCP Support

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Authentication By specifying authentication settings on REGISTER or INVITE requests, you can limit calls that go through Brekeke SIP Server. Authentication Plug-in is available for the users who wish to use an existing user directory service. Authentication Plug-in Sample

Upper/Thru Registration Upper/Thru Registration, the Brekeke SIP Server's original function, allows for easy configuration of parallel users of pre-existing or other SIP servers. With this feature, you can take advantage of SIP communication through ITSP lines or third party SIP Servers. Upper/ThruRegistrationFeature

Session Management The real-time session management is available through Brekeke SIP Server's Administrative tool. View session status ormanuallyterminatethe active calls.

Multiple-Domain Hosting Brekeke SIP Server can host multiple domains on one server install. This feature allows user to manage

multiple domains under one server setup.

SIP express router (SER) Standards SERi complies to the SIP standard family. SER has been extensively tested during many SIP interoperability events. Due to the fact that it is free SER is being used by many SIP device vendors internally, this helps us to achieve excellent interoperability. Continued SER development focus on developing new functionality that is well-implemented and is useful to users of SER. The focus of SER is primarily on large-scale (i.e. service provider, enterprise) setups and other setups requiring the same type of reliability.

Presence (presence events package RFC3856, SIP SUB/NOT RFC3265, pidf RFC3863, PUBLISH RFC3903, presence.winfo RFC3857, presence.winfo data format RFC3858, authorization draftietf-geopriv-common-policy-05 & draft-ietf-simple-presencerules-03, resource lists draftietf-simple-xcap-list-usage-05 & draft-ietf-simple-event-list-07) CPLi RFC 3880 Note that SER as a proxy server supports many other SIP methods, payload types and header-fields "transparently" by just forwarding requests with unknown elements in it blindly but correctly to the right place. Prominent examples of methods supported transparently include but are not limited to INFO method (RFC2976), ISUP/QSIP MIME Types (RFC3204), UPDATE method (RFC3311), PRACK method (RFC3262), MESSAGE method (RFC3428), Replaces header field (RFC3891), Join header field (RFC3911), etc.

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Supportedstandardsinclude: SIP (RFC3261) and referred protocols (TLS/RFC2246, digest/RFC2617, DNS, URIi/RFC2396, telURI/RFC3966) Locating SIP Servers (RFC3263) ENUM (RFC 3761) Asserted Identity RFC 3325 (configuration feature) Messaging (RFC 3428) Symmetric Responses RFC 3581 Session Timers RFC 4028 (configuration feature) Connection Reuse (draft-ietfsip-connect-reuse)

Network Integration Building networks is harder than just starting the SER software. You need to integrate the running server with clients behind NATs, coordinate numbering plans with rest of the world, ensure proper capacity, etc. SER can be configured to serve in all such scenarios: it has build-in loadbalancing support, NATiitraversal,ENUM.

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Security

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SER comes with built-in security features for detecting abnormal conditions (pike, ratelimit) and enforcing Access Control Lists (permissions, group modules)

Underthe Hood With the SER code being written in plain C, portability is excellent, footprint is small, and speed is high Modules-API allows introduction of new functionality without affecting existing codebase Transport protocols supported: IPv4/v6, UDP, TCP, TLS

pua -- command line presence UA o SER command-line tools: ser_* (ser_ctl, ser_cred, ser_rpc, ser_user, ser_attr, ser_domain, ser_uri) Embedded
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milkfish -- home router SIP connection tracking restund -- STUN/TURN server

NATs
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PSTNi Asterisk -- popular open source PSTN gateway Security
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3rd Party SIP Software Key free SIP components maintained by iptel.org include: SER -- SIP Express Router, RFC3261 SIP proxy SEMS -- SIP Express Media Server SERweb -- web interface toSERi RTP Proxy -- media relay for NATi traversal (courtesy of Maxim Sobolev) SER command-line tools: seri_* (ser_ctl, ser_cred, ser_rpc, ser_user, ser_attr, ser_domain, ser_uri) mystun -- STUN server Besides that, many third-party free tools have been reported to be used along with iptel.org SIP software. Accounting CDR Tooltool CDRipostprocessing o OSP Toolkit Command-line tools
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sipcracktoolfordictionary attacks o sipbot o smapport scanner o protos buffer overflowtestingtool o Tactical VoIP toolkit VoIPi security assessment tool o see also MohamenNassar's list of further security tools, and VOIPSA's security tool list. Stacks
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libre -SIP, RTPi/RTCPi, STUN/TURN/ICE, BFCP and DNS o sofia-sip o pjsip Testing
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SDPi,

for

sipp performance stress tool o seagull -IMSitrafficgenerator Troubleshooting
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ethereal capturing

tool VoIP

for (and

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UA
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other) traffic and analyzing it routegraph -- creates call graphs for SER configuration files SIP Analyzer SIP v6 analyzer sipgrep -- filtering of SIP messages sipsaktraceroute-like tool for SIP (with many other features); also easy to integrate into monitoring applications such as argus, nagios sipscenario tool for generating HTML-ized traces of SIP messages sipspy -tool for capturing and graphical representation of SIP traffic in multiple nodes wist -remote SIP tracing see also Klaus Darilion'slist of SIP testing tools baresip -- command line high quality voice + video softphone jitsi -- (formerly SIPCommunicator) cross platform voice + IM + video Java client kphone minisip -- SIP UA Twinkle

The ATA 188 adds a switch with a second ethernet port. These devices convert two analog phones to IP phones and support the following protocols: SIP H.323 MGCP SCCP

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If you purchased the ATA new (not though a VOIP provider,) you get at least one free support call. If lucky and you press them enough, they will give you access to the SIP firmware for download. If you go the contract route, the SmartNet package that applies is part #CON-SNT-ATA186 for the 186 and CON-SNT-ATA188 for the 188. TheysupportthefollowingCodecs: G.729, G.729A, G.729AB2 G.723.1 G.711a-law G.711u-law (Note: The ATA only supports G.729A on one channel at a time due to CPU load. Theotherchannelwill use G.711)

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Tomadode : http://www.iptel.org/3rdpsip/ http://www.iptel.org/ser/features

ATA Allwin h100 series

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Cisco ATA188

specifications • Physical: 104 x84 x27 mm (W/D/H) • Power: 12V DC, 1.0A Adaptor • Weight: 110g • WAN: 10/ 100M bps RJ-45 connector,(Auto crossover) • LAN: 1-port 10/ 100M bps Ethernet

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Switch. (Auto crossover) • VoIP: 1 SLIC(FXS) connect to phone set • PSTN: 1FXO port for PSTN line • LED Indicators : Power, Ready, Status, WAN linking , LAN linking, VoIP Line, PSTN Line • Supported Protocol: UDP, TCP, Standard SIP/H.323,NAT,BOOTP, TFTP, FTP, HTTP, TELNET, IEEE 802.3/ IEEE 802.3u • Selectable Coders: G.711, G.723.1, G.726 , G .729a b DTMF / Call progress tone detection • and generation • G.168 echo cancellation 1 Reset button for load factory • default IP parameters setting • User friendly Web configure interface Configuration/ Upgrade by FTP and • Web Build-in watching dog for auto • recovery

http://www.voipinfo.org/wiki/view/Analog+Telephone+Adapters

ATA YGW30 YGW30 1FXS,1FXO SIP ATA

Support T.38(Doing) IP/TCP/UDP/RTP/RTCP IP/ICMP/ARP/RARP/SNTP TFTP Client/DHCP Client/ PPPoE Client Telnet/HTTP Server DNS Client NAT/DHCP Server Support G.723.1 5.3k/6.3k,G.729,G.711 A-Law、μLaw audio codec algorithm Dynamic voice detection Echo cancellation Comfort noise generation Tone generation and Local DTMF generation and detection according with ITU-T Settings by HTTP web browser (IE6.0) Advanced settings by Telnet Voice prompt Upgrade by TFTP 2RJ45 Ports,Built-in Router,conference,Auto-provison or updating by HTTP,FTP or TFTP. For each YGW30A,it can have 3-SIP account,all can be used as caller at anytime.It can be selected among these 3-SIP account as caller by dial different dial prefix.It can be used as PSTN ordinary phone when power off. For each YGW30B,it can have 3-SIP account and one PSTN phone number,that means each ATA own 4 phone numbers,all can be used as callee at anytime.For the caller,these 4 account can be selected by dial different relevant prefix(including switch to PSTN as an ordinary PSTN phone) YGW30B has a real FXO port to support router call from PSTN to VOIP or from VOIP to PSTN.

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Performance and features SIP v1 (RFC2543), v2(RFC3261) Support Route,Two 10/100Mbps MACs

Con formato: Fuente: +Títulos, 11 pto, Inglés (Estados Unidos)

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