Windows media player rtcpemplois

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    86 windows media player rtcp travaux trouvés au tarif de EUR
    PJSUA PROBLEM 3 heures left

    ...Call 0: initializing media… 17:21:01.636 pjsua_media.c !..RTP socket reachable at [se connecter pour voir l'URL] 17:21:01.636 pjsua_media.c …RTCP socket reachable at [se connecter pour voir l'URL] 17:21:01.636 pjlib …Failed to register socket to ioqueue because socket fd is too big (fd=1854/FD_SETSIZE=1024) 17:21:01.637 pjsua_media.c …Unable t...

    €50 (Avg Bid)
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    1 offres
    Configure OPENSIPS S'est terminé left

    1: Functional characteristics - Individual RTP media streams shall use an associated RTCP channel to report transmission performance. - The media transported in an individual RTP media stream shall be as defined in the associated SDP description carried over the SIP signalling link. - The SIP signalling shall carry embedded ISUP signalling in accordance

    €166 (Avg Bid)
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    2 offres

    Mandatory Skills: 1) Hands on expertise in SIP, RTP and RTCP 2) Good knowledge on TUN,STUN, NAT 3) Free-switch working knowledge on audio conference using SIP and Web RTC 4) Good knowledge on RTP Proxy and routed audio conferences concept where media would flow via free switch RTP Proxy 5) Working experience of High Availability and Cluster 6) SDP

    €29 / hr (Avg Bid)
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    1 offres

    The VMS platform project should have a universal form of support for mjpeg H264 formats.h264+. H265+ support IPv4 / IPv6, TCP, UDP, RTP, RTSP, RTCP, HTTP, HTTPS, DNS, DDNS, DHCP, FTP, NTP, SMTP, UPnP, SIP, SNMP, PPPoE, VLAN, 802.1 x, QoS, ONVIF Profile S & G, and to integrate via the SDK P2P, video Analytics modules [se connecter pour voir l'URL]

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    9 offres

    I need to sync rtp with caller and called with RTCP SR. Project must be done in c# . SSRC and Payload changes. Codecs are G711,G729,AMR WB,AMR NB,EVS.

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    4 offres
    IP Camera Streaming S'est terminé left
    VERIFIÉ

    ...to stream from IP Camera to wowza server. IP Camera Specification: Wifi Supported Camera. Support of Network Protocol - TCP/IP,ICMP,HTTP,HTTPS,FTP,DHCP,DNS,DDNS,RTP, RTSP,RTCP,NTP,SMTP Work Out : - Power On the IP Camera - IP camera should be connected to Wifi Hot Spot - One time setup of provided application or some settings to stream to

    €514 (Avg Bid)
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    10 offres
    Design me a Product S'est terminé left

    i want make scribt controll all rtp and rtcp i want control all this option i use freepbx Got RTCP from my ip :45403 PT: 200(Sender Report) Reception reports: 0 SSRC of sender: 1419880292 NTP timestamp: 3693388389.0308424704 RTP timestamp: 2515039040 SPC: 200 SOC: 8955 Our SSRC: 151061569 Their SSRC: 1688707412 iFraction lost: 0 Cumulative

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    5 offres
    freepbx with Webrtc solution S'est terminé left

    ...dtls_srtp_setup: Could not set policies when setting up DTLS-SRTP on '0x7fd4bc006ea0' [2016-12-28 15:11:11] WARNING[28418][C-00000006]: res_rtp_asterisk.c:4120 ast_rtcp_read: RTCP Read error: Unspecified. Hanging up. if you are 100 % you can solve it ... bid else , don't bid plz , no try and error im willing to pay 100 $ for solution

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    7 offres

    ...n se valora experiencia en Amazon Web Services, message queue's (ZeroMQ, ActiveMQ, etc.), escalabilidad y protocolos de streaming de audio / video (ejemplo RTSP, RTMP, RTP, RTCP, SIP)<br /> <br /><br />Duración del proyecto: Larga duración<br />Roles a cubrir por el freelance: Desarrollador<br />Compatibilidad con: Linux / Unix<br ...

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    1 offres

    ...Book AND the platform 4. User Credentials Save / Change Username & Password 5. Volume Control 6. Protocol SIP (RFC 3261) Integration with UI 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

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    24 offres

    ...Book AND the platform 4. User Credentials Save / Change Username & Password 5. Volume Control 6. Protocol SIP (RFC 3261) Integration with UI 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

    €2511 (Avg Bid)
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    51 offres

    Requirements: Extensive Knowledge of VoIP and SIP protocol Background of Open source PBX (Asterisk/Freeswitch) Experience with streaming protocols including RTP, RTCP, RTSP, RTMP and HTTP Expert knowledge of Java (Spring Framework, Asterisk-Java) Knowledge of web services and Tomcat applications

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    5 offres
    Build a Website S'est terminé left

    Requirements: Extensive Knowledge of VoIP and SIP protocol Background of Open source PBX (Asterisk/Freeswitch) Experience with streaming protocols including RTP, RTCP, RTSP, RTMP and HTTP Expert knowledge of Java (Spring Framework, Asterisk-Java) Knowledge of web services and Tomcat applications

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    LINPHONE / OPENFIRE EXPERT S'est terminé left

    ...linphone and liblinphone and openfire. Skills required: - sip (for audio and video calls) using linphone - xmpp (for presence and messaging) - adaptive bit rate throught rtcp packets Before answering for the ad write "I KNOW LINPHONE". Also in your application state where you used linphone Failure to do both will get your application rejected

    €2099 (Avg Bid)
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    VOIP DIALER, SIP DIALER S'est terminé left

    ...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

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    1 offres

    ...Book AND the platform 4. User Credentials Save / Change Username & Password 5. Volume Control 6. Protocol SIP (RFC 3261) Integration with UI 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

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    9 offres

    ...Book AND the platform 4. User Credentials Save / Change Username & Password 5. Volume Control 6. Protocol SIP (RFC 3261) Integration with UI 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

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    Desarrollar una aplicación en Android para mostrar canales IPTV (UDP, RTCP), obteniendo lista de canales a través protocolo SAP.

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    3 offres
    Android voip softphone S'est terminé left

    ...Address Book Integration with local phone 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

    €604 (Avg Bid)
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    12 offres
    VOIP DIALER, SIP DIALER S'est terminé left

    ...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

    €84 (Avg Bid)
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    2 offres
    Android/iPhone SIP Dialer S'est terminé left

    ...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

    €2380 (Avg Bid)
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    21 offres
    VoIP SIP Intercom Module S'est terminé left

    We are planning for Embedded Small size Module for VoIP intercom support SIP,RTP,Codecs, AEC,Jitter,FEC,RTCP,.... Ethernet, Serial, GPIO, Audio in/out,MIC, base on Linux and opensource platform The main goal are Stability Benchmark Compatibility Functionality Similar products is [se connecter pour voir l'URL]

    €1582 (Avg Bid)
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    10 offres

    ...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

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    17 offres

    ...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

    €1938 (Avg Bid)
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    21 offres
    VOIP DIALER, SIP DIALER -- 2 S'est terminé left

    ...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

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    12 offres
    VOIP DIALER, SIP DIALER S'est terminé left

    ...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

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    9 offres
    VOIP DIALER, SIP DIALER S'est terminé left

    ...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

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    6 offres
    SIP Dialer S'est terminé left

    ...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 [se connecter pour voir l'URL] G.729, AMR (Adaptive Mult...

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    I need a script to do the fallowing : I have a list with names . Example Maria Barto...bartoni[at][se connecter pour voir l'URL] [se connecter pour voir l'URL][at][se connecter pour voir l'URL] etc . Will combine all names and will create all possibles of emails after this your script should validate the emails what he create by RTCP or any other method to check ...

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    ...with various trans-coding protocol to use based on Least Cost Route for Inbound and outbound Traffic. Telephony/VoIP/IP protocols like :SS7, SIP, SIP-I/SIP-T, H323, ENUM, RTP, RTCP, ISDN, IPSec. VoIP Configuration and Programming FreeSwitch Installation & Configuration SBC and OpenSip Asterisk and OpenVPN for Transcoding and Routing Server with Codecs

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    8 offres

    ...with various trans-coding protocol to use based on Least Cost Route for Inbound and outbound Traffic. Telephony/VoIP/IP protocols like :SS7, SIP, SIP-I/SIP-T, H323, ENUM, RTP, RTCP, ISDN, IPSec. VoIP Configuration and Programming FreeSwitch Installation & Configuration SBC and OpenSip Asterisk and OpenVPN for Transcoding and Routing Server with Codecs

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    1 offres
    VOIP DIALER, SIP DIALER S'est terminé left

    ...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

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    23 offres
    VOIP DIALER, SIP DIALER S'est terminé left

    ...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

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    24 offres

    ...backups and system security. Skills & Specifications: • Strong engineering and operational experience in service provider networks. • In-depth understanding of SIP, RTP and RTCP • Experience configuring and maintaining open source SIP server software including patching, upgrades, logs, debugs and virtual traces. • In-depth understanding of open source

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    ...Analysis, Design and Development of full life cycle internet video projects. Develop applications that control, manage and process media on Linux and/or Windows-based x86 platforms. Applications include soft transcoding and media re-wrapping, or real-time encode/ingest and decode/play. Requires extensive experience in programming and good design skills with

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    VOIP DIALER, SIP DIALER S'est terminé left

    ...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

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    22 offres
    Android/iPhone SIP Dialer S'est terminé left

    ...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

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    SIP Dialer S'est terminé left

    ...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 [se connecter pour voir l'URL] G.729, AMR (Adaptive Mult...

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    Android/iPhone SIP Dialer S'est terminé left

    ...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

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    Call Egypt S'est terminé left

    ...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

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    25 offres
    VOIP DIALER, SIP DIALER S'est terminé left

    ...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

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    24 offres
    Android/iPhone SIP Dialer S'est terminé left

    ...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

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    communication (call audio/ vedio) android phone using sip nad rtpou rtcp with code source

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    We need to have some Server Code developed in Java for an audio distribution server that received data via RTP/RTCP and distributes data via RTP/RTCP A detailed Spec (detailed solution architecture and technology recommendation; some UML class descriptions) and a functional spec is available. We have also defined the interface to the client.

    €662 - €1324
    À la une Scellé
    €662 - €1324
    10 offres

    ...24.0.1312) WebRTC media stack: VP8, iSAC, BUNDLE, rtcp-mux, STUN, ice Able to place and receive SIP calls Simple UI (only need a spot to enter the SIP URI, place call, hang up, answer) Windowed and full screen video We will provide: Centos 6.3 linux server for testing git repo for project Websocket to SIP gateway Media Server that you can

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    I need someone to develop me a very simple Windows C++ application that uses the webrtc library to provide real time voice and video communications between two PCs. The requirements are: 1) C++ application must compile using Visual Studio Express 2) Application must run on Windows 7 and Windows 8 3) Application must use webrtc and dependencies

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