搜索资源列表
VoiceEngines
- 这是一个将语音打包程序, 将数据打包成RTP/RTCP-This is a package of voice procedures, data packing into RTP / RTCP
jmf-2_1_1e-scsl-src
- JMF implement the multimedia communiacation, it will endecoode the voice input to rtp package and send through the network.-JMF implement the multimedia communiacat ion. it will endecoode the voice input to rtp package and send through the network.
javasoundsend
- 关于JAVA多点对多点的语音传输问题,采用rtp与rtcp实现语音的传输-on JAVA more point-to-multipoint transmission of voice, the problem with RTCP rtp voice transmission
jvoiplib-1.4.1
- 这是一个外国人写的语音聊天代码,其中使用了RTP协议-This is a foreigner's voice chat write code, which used the RTP
rtpvoice.rar
- RTP voice transmission with real time protocol of jmf,RTP voice transmission with real time protocol of jmf
PacketToWave.rar
- 实现对RTP包的语音提取，可以直接听取RTP包里的语音流。,RTP voice packets to realize extraction
MChat.rar
- 一个可以在局域网进行视频聊天的源代码，语音压缩采用G729，视频压缩采用H.263，网络传输采用RTP。本程序介绍了视频聊天的基本技术，稍加改动就可以直接运用于Internet网络。,A local area network for video chat in the source code, voice compression using G729, video compression using H.263, network transmission using RTP. This proc
pro
- 本技术文档是本人再学习过程中自己翻译理解后的总结。关于Socket、SIP、RTP、G729、ARM语音编解码、H263、H264视频编解码。包括理解，基本的内容和算法等等。-The technical documentation is a learning process, I no longer understand their own after the translation of the summary. On the Socket, SIP, RTP, G729, ARM voice
JMF-RTP
- 基于JMF RTP的语音视频传输，比较完善，具体的下载者可以自己修改-JMF RTP-based voice and video transmission
thesis-jori-original
- 国外牛人的毕业论文，英文版本，实时语音传输相关，其中的两个库文件，jrtplib与jvoiplib已经上传。enjoy -foreign elite s graduate paper in english.related to real time voice transimission.it has two library,jrtplib,jvoiplib.enjoy
PacketToWave
- 实现从RTP中分析和提取语音。可以直接播放或者存成wav文件-From RTP voice in the analysis and extraction. Player can be directly or saved as wav file
udp_linux.tar
- sip软电话，能够基于rtp协议实现简单的语音传输。-sip soft phone, can rtp protocol implementation based on a simple voice transmission.
gsm
- 这个网络电话程序是linux下，用C语言实现的。它既不是实现的H.323 或 SIP协议, 也没有使用RTP协议，更没有使用到任何其它第三方软件，不过，它确实工作的很好。通话话音质量非常不错。-The network telephone program is under linux, using the C language. It is not the realization of the H.323 or SIP protocol, but also did not use the RTP
658jrtplibmediaplayer
- 利用rtp库实现实时语音传送，是做voip的必备协议-Rtp library using real-time voice transmission, is so essential agreement voip
VS2005sipcode
- SIP它既不是会话描述协议，也不提供会议控制功能。为了描述消息内容的负载情况和特点，SIP 使用 Internet 的会话描述协议 (SDP) 来描述终端设备的特点。SIP 自身也不提供服务质量 (QoS)，它与负责语音质量的资源保留设置协议 (RSVP) 互操作。它还与若干个其他协议进行协作，包括负责定位的轻型目录访问协议 (LDAP)、负责身份验证的远程身份验证拨入用户服务 (RADIUS) 以及负责实时传输的 RTP 等多个协议。-It is not a SIP Session Descr
DealwithData
- using G.729 language real-time communication [G729b_v14Code.rar] - G729B the realization code, and G729A difference is he could support multithreading codec, multi-channel audio simultaneously decode [MChat.rar] - a LAN can video chat source co
andreadrian.deintercom
- Voice over IP Intercom Linux 环境下，带有回声消除模块，VC++开发。-The application can: dial an intercom partner via short-dial buttons transport your voice over IPv4 with RTP UDP unicast data packets make a telephone conference support wideband (16kH
IPwangluoyuyin
- 实现网络在线语音对话.支持GSM编码,ADPCM编码,LPC编码,LPC-10编码 支持RTP,vat协议,并有广播发送,按组进行多点传送,文字交谈等功能-Networked online voice dialogue. Support GSM encoding, ADPCM coding, LPC coding, LPC-10 encoding support for RTP, vat agreement, and announcements are sent, according to
mediastreamer
- Mediastreamer开发手册 Mediastreamer is a library written in C that allows you to create and run audio and video streams. It is designed for any kind of voice over IP applications. It features RTP connectivity, audio codecs (Speex, iLBC, G711, GSM), vi
6_p2pAudio
- 文件名 aocx2.ocx， P2P 语音传输模块，可以穿越 100 防火墙，其中有 80 数据不通过服务器，极大减轻服务器压力，RTP协议，所需带宽 2KB，延迟200ms以下。语音清晰，可商用。 可用于ASP, JSP, .NET, VB, Delphi, Vc++, C++Builder, PB等各种开发语言的程序中。 -File Name aocx2.ocx, P2P voice transmission module, can pass through the fire