marketo rtp

DTMF definition: Digital keys (0 ~ 9 * # a B C D ). There are usually three methods for detecting DTMF in VoIP: SIP info, inband, and out band (rfc2833). In addition, the latest RFC has been adopted for the requirements of DTMF In the 3GPP IMS specification.4733 replaces RFC 2833. 1. Sip info For out-of-band detection, DTMF Data is transmitted through the sip signaling channel. There is no unified implementation standard, which is sent through the SIP info method. The signal field in the packa

. 3. Does a NALU correspond to a piece?This statement is not accurate. NALU includes a piece, SPS, PPS, SEI, etc. 4. decode_one_frame () includes I, P, and B. 5. Case nalu_type_slice: Case nalu_type_idr: Case nalu_type_dpa Case nalu_type_dpb: Case nalu_type_dpc Case nalu_type_sei: Case nalu_type_pps Case nalu_type_sps Case nalu_type_aud: Case nalu_type_eoseq: Case nalu_type_eostream: Case nalu_type_fill Question: When to enter and what is the descriptionArticleOr books? A

Options rtsp: // 192.168.1.154: 8557/h264 RTSP/1.0CSeq: 1User-Agent: VLC Media Player (live555 streaming media v2010.05.28)RTSP/1.0 200 OKCSeq: 1Date: sat, Jan 01 2000 00:05:11 GMTPublic: Options, describe, setup, teardown, play, pauseDescribe rtsp: // 192.168.1.154: 8557/h264 RTSP/1.0CSeq: 2Accept: Application/SDPUser-Agent: VLC Media Player (live555 streaming media v2010.05.28)RTSP/1.0 200 OKCSeq: 2Date: sat, Jan 01 2000 00:05:11 GMTContent-base: rtsp: // 192.168.1.154: 8557/h264/Content-Type:

. So although the TS Stream format is in MEPG-2 Defined, but it can also be used to pass the media file of the MEPG-4, just because it is defined in the MPEG-2, so it is often called the MEPG-2 TS stream. In terms of media processing methods, from the encoding end to the decoding end, we need to establish multiple RTP for audio and video and other data streams. Session. Therefore, when I/O is used as a streaming media server, you need to manage multip

= CreateMediaSession(streamName);}if (!m_mediaSession){handleCmd_notFound(); return -1;}MediaSession * session = m_mediaSession;std::string sdp = session->GenerateSDPDescription(m_serveAddr);//get the rtsp url//rtsp://127.0.0.1/std::string rtspUrl;append(rtspUrl, "rtsp://%s:%u/%s", m_serveAddr._ipstr(),m_serveAddr._port() ,session->StreamName().c_str());std::string response = "RTSP/1.0 200 OK\r\n";append(response, "CSeq: %u\r\n" "%s" "Content-Base: %s\r\n" "Content-Type: application/sdp\r\n" "

1, in the NS simulation network, the Grouping (Packet) is the basic unit of interaction between objects. A grouping is a series of grouping headers and an optional data space composition. The structure of the packet header is initialized when the simulator object is created, and the offset of each packet header relative to the starting address of the packet is also recorded. By default, most NS built-in packet headers are enabled (including common headers, IP headers, TCP headers,

First, an overview of the WEBRTC audio processing flow, see:WEBRTC The audio session is abstracted into a channel channels, such as A and b for audio calls, a needs to establish a channel and b for audio data transmission. There are three channel, each channel contains codec and rtp/rtcp send function.In the case of a channel, the application will contain three active threads, a recording thread, an audio receive thread, and a playback thread.1) Recor

real-time audio and video domain UDP is the king In the Internet, audio and video real-time interaction using the Transport Layer Scheme has TCP (such as: RTMP) and UDP (such as: RTP) two kinds. The TCP protocol can provide a relatively reliable guarantee for data transmission between two endpoints, which is achieved through a handshake mechanism. When the data is passed to the receiver, the receiver checks the correctness of the data. The sender can

Real-time Streaming protocol RTSP (Realtimestreamingprotocol) is proposed by RealNetworks and Netscape, which defines how a one-to-many application can efficiently transfer multimedia data over an IP network. RTSP is on the architecture of RTP and RTCP, which uses TCP or RTP to complete data transfer. HTTP transmits HTML compared to RTSP, while RTP transmits mult

reduces bandwidth and improves response times by caching and reusing frequently-requested Web pages. Squid has extensive access controls and makes a great server accelerator. It runs on most available operating systems, including Windows and is licensed under the GNU GPL.Features:Making the most your Internet ConnectionWebsite Content Acceleration and distributionVarnishVarnish is a state-of-the-art, high-performance HTTP accelerator. It uses the advanced features in Linux 2.6, the FreeBSD 6/7

, UDP may cause packet loss. To avoid this, you can add the-S parameter (for example,-s 320x240) to reduce the resolution.2.4. encoding: H.264, released RTP The following command is used to obtain the camera data-> encode it as H.264-> encapsulate it as RTP and send it to the multicast address. ffmpeg -f dshow -i video="Integrated Camera" -vcodec libx264 -preset:v ultrafast -tune:v zerolatency -f

2008.7.17 The program cannot connect to the server through cmnet on the mobile phone, but the mobile phone can connect to the server (such as HTTP and RealPlayer ). You can also connect to the Internet IP address of the server in the simulator. You have to study rtspclient and socket in the morning. 2008.7.18 Test with rsocket example. If the problem is the same, the connection cmnet prompts "conn. Failed-46". The permission is insufficient. Originally, the socket required the networkservices ca

requires a lot of overhead, it is not suitable for transmitting real-time data. In the implementation scheme of stream transmission, HTTP/TCP is generally used to transmit control information, however, RTP/UDP is used to transmit real-time sound data.The process of stream transmission is usually as follows:(1) After selecting a first-class media service, the web browser and the Web server use HTTP/tcp to exchange control information so that the real-

variety of operating systems, including Windows and Linux. it has full support for UDP, TCP, and TLS transports on both IPv4 andIPv6. It also implements the full set of specifications for DNS usage in SIP, including naptr and SRV lookups (rfcs: 3263,291 5, 2782) using an asynchronous DNS Library (Ares ). The resiprocate project consists of a stack and a small collection of applications. the resiprocate stack is currently used in two specified cial products and is quite stable. resiprocate is id

Options rtsp: // 192.168.1.154: 8557/h264 RTSP/1.0CSeq: 1User-Agent: VLC Media Player (live555 streaming media v2010.05.28)RTSP/1.0 200 OKCSeq: 1Date: sat, Jan 01 2000 00:05:11 GMTPublic: Options, describe, setup, teardown, play, pauseDescribe rtsp: // 192.168.1.154: 8557/h264 RTSP/1.0CSeq: 2Accept: Application/SDPUser-Agent: VLC Media Player (live555 streaming media v2010.05.28)RTSP/1.0 200 OKCSeq: 2Date: sat, Jan 01 2000 00:05:11 GMTContent-base: rtsp: // 192.168.1.154: 8557/h264/Content-Type: