rtp careers

, and other parameters. The initialization information is not directly applied to devices, but stored in the structure type parameter pctx of an encoding parameter. Then, you can use the following code to set parameters, that is, to apply the parameters to actual devices.IOCTL (pctx-> hopen, cmd_init, mfc_args );The encoding part is implemented using the next line of code.Ioetl (pctx-> hopen, cmd_exe, mfc_args );After encoding, you can use the function to obtain the memory address of the encod

Transferred from: http://tieba.baidu.com/p/2138076570Abstract: In order to solve the problem of audio and video synchronization caused by delay, jitter and network transmission conditions, a new audio and video synchronization scheme is designed and implemented to adapt to different network conditions. Using the audio and video coding technology, AMR-WB and H. e have the characteristics of rate selectable in complex network environment, combine RTP ti

signal.For out-of-band detection, a DTMF signal is carried through the info method of the SIP signaling. There is no uniform implementation standard, and the DTMF keys are identified by the signal field in the Sipinfo package with the Cisco sipinfo standard . Note that when DTMF is "*" The different standards implement the corresponding signal=* or signal=10. The advantage of Sipinfo is that it does not affect the transmission of RTP packets, but b

through coaxial cable, such as transmission, IP network development, the use of IP network excellent transmissionrelating to technology or agreement:Transport protocol: RTP and RTCP, RTSP, RTMP, HTTP, HLS (HTTP Live streaming), etc.Control signaling: SIP and SDP, SNMP, etc.4, decoding the data:Using the relevant hardware or software to decode the received encoded audio and video data and get the image/sound that can be directly displayedrelating to t

variable setting value is very large case. The switch can be switched on and off directly with the OK key. Finally, also when you are free to manipulate character activities, if you hold down the CTRL key to move, you can ignore the map tile's traffic, even on the inaccessible components can also move freely. You can use this when the game map is large and you only want to get to the point where the plot is triggered, or if you accidentally have an unacceptable block of traffic. But if it is th

sampled 64 kbit/s voice to 8 kbit/s with almost no loss of quality. Because the service quality in the group switching network cannot be well guaranteed, the voice encoding must be flexible, that is, the variable encoding speed and the variable encoding scale. G.729 was originally the 8 kbit/s voice encoding standard, and now the scope of work is extended to 6.4 ~ 11.8 kbit/s, the voice quality has also changed in this range, but even 6.4 kbit/s, the voice quality is also good, so it is very su

Tunneling RTSP in HTTP Status of this memo This document is an internet-Draft and is in full conformanceAll provisions of section 10 of rfc2026. Internet-drafts are working events of the Internet EngineeringTask Force (IETF), its areas, and its working groups. Note thatOther groups may also distribute working clients as Internet-Drafts. Internet-drafts are draft documents valid for a maximumSix months and may be updated, replaced, or obsoleted by otherEvents at any time. It is inappropria

terminal to prevent the echo from being encoded into the voice stream, so as to improve the Speech Quality of the peer experience. 2.3 Jitter: Jitter buffer is generally used to solve the jitter problem, but this solution will increase the system delay time. To achieve the best effect, the jitter buffer size should be dynamically adjusted according to the actual jitter; The jitter level can be determined based on the RTP timestamp (CiscoIOS ); 2.4

Http://blog.chinaunix.net/uid-15063109-id-4445165.html————————————————————————————————————————————————————————————Pjsip function is very strong, do SIP RTP voice call Library preferred. After 2.0, video is also supported. However, its video function is collected from the video device by default, then compiled and sent out. Suppose we already have a video source, such as an IP camera, and do not need to collect and encode this process, how to deal with

This page is maintained by Paul E. Jones. Core sip documents RFC 2543 SIP: Session Initiation Protocol (obsolete) RFC 3261 SIP: Session Initiation Protocol SDP Related Documents RFC 2327 Session Description Protocol (SDP) RFC 3264 An offer/Answer Model with the Session Description Protocol (SDP) RFC 3266 Support of IPv6 in SDP RFC 3388 Grouping media lines in

3.2.21 RTPSession --------------------------------------------------------------------------- Header file: rtpsession. h For most RTP-based applications, the RTPSession class may be the only one that needs to be used. It automatically processes the RTCP part internally. Therefore, you can focus on sending and receiving actual data. Note: The RTPSession class does not mean thread security. You need to use the locking mechanism to prevent different thre

a film?This is not a very accurate statement, Nalu includes a film, SPS, PPS, Sei and so on4, Decode_one_frame () including I, P, B5. Case Nalu_type_slice:Case NALU_TYPE_IDR:Case NALU_TYPE_DPACase NALU_TYPE_DPB:Case NALU_TYPE_DPCCase Nalu_type_sei:Case Nalu_type_ppsCase Nalu_type_spsCase Nalu_type_aud:Case NALU_TYPE_EOSEQ:Case Nalu_type_eostream:Case Nalu_type_fillQuestion: When to enter which, what is the description of the article or book?A: Which case to enter is determined by the Nalu_type

Pjsip function is very strong, do SIP RTP voice call Library preferred. After 2.0, video is also supported. However, its video function is collected from the video device by default, then compiled and sent out. Suppose we already have a video source, such as an IP camera, and do not need to collect and encode this process, how to deal with it? Let's say we use the Pjsua included with Pjsip as an example.The usual method:1 The video source of course fi

stream play and is a pure virtual class. where Startstream and getstreamparameter are pure virtual functions. ondemandservermediasubsession: Added a streamsource processing and rtpsink handles functions and classic named properties. Package seek,pause and other processing, these interfaces Clientsessionid to here converted into framedsource. the member functions of the class most and Servermediasubsession similar, in the streaming media complete positioning processing. createnewstrea

(siren7), g.723.1, g.726, g.728, g.729a;Stereo codec (L16 );Wav file playback, streaming media and recording;Supports the RTCP protocol;Call quality monitoring;RFC 2833;Automatic response, automatic playback of files, automatic cycle of RTP;Generate sound;AEC (accoustic echo cancellation );Adaptive jitter buffer;Adaptive mute detection;PLC (package loss and hiding );Packet loss simulation;Each RTP packet c

(1) SDP description format(2) SDP example(3) SDP(1) SDP description formatM=video 1234 RTP/AVP 96a=rtpmap:96 H264A=framerate:15C=in IP4 192.168.0.104Above is a self-written RTPM=audio 1234 RTP/AVP 0a=rtpmap:0 PCMA/8000/1A=framerate:25C=in IP4 172.18.168.451.m= is the beginning of a media-level session, Audio: media type; 1234: port number; RTP/AVP: transport prot

upgraded to support multi-directory and multi-vendor connectivityA Conversion format for rfc2279 UTF-8, ISO 10646RFC2281 Cisco hot backup routing protocol (HSRP)Multi-Protocol extension for rfc2283 BGP-4Rfc2284 PPP scalable Authentication ProtocolRfc2289 one-time password systemRfc2296 HTTP remote variable selection algorithm-rvsa/1.0Rfc2313 PKCS #1: RSA encrypted version 1.5Rfc2330 IP address execution rule managementRfc2343 is applied to the format of the bound mpeg

= * (connection information-this field is not required if it is included in all media)B = * (bandwidth information)One or more time descriptions (as shown below) Z = * (Time Zone adjustment)K = * (encryption key)A = * (0 or multiple session attribute rows)0 or more media descriptions (as shown below) 2. Time description T = (Session Activity time)R = * (0 or repeated times)3. Media description M = (media name and transfer address)I = * (media title)C = * (connection information-this field i

, and sensitive to transmission latency and jitter. However, under certain circumstances, packet loss can be allowed, that is, a certain degree of Transmission Error code is acceptable. In addition, the streaming media service must meet the needs of broadcast and multicast applications, and must have the ability to adjust the video transmission quality according to the real-time available transmission bandwidth of the network. To provide streaming media data services over the Internet, you must

.10.1.1.1: 123 --- Nat ---> 202.70.65.78: 10000 ------ PC (B)If PC (B) also sends data to 202.70.65.78: 10000, the data is sent to 10.1.1.1: 123.5. Restricted cone Nat of port restricted portIn addition to the four conditions, it is necessary not only to check the source IP address of PC (a), but also to check whether its port is the same as the previous one.10.1.1.1: 123 ---> Nat ----> 202.70.65.78: 10000 -----> PC (a) [213.123.324.34: 8000]This NAT will only receive data from the IP address 21