marketo rtp

| "551"; Option not supported| Extension-codeExtension-code = 3DIGITReason-phrase = *Example The following is the RTSP negotiation process for a province IPTV: request Media URL Accept:application/sdpNegotiated to describe the media information protocol Cseq:1 User-agent:zte ltd.co RTSP protocal verion 1.0 guid-2.1.1.100/b519d290-c0ec-ee35-7368-893be4c0b347 User agnet information, display is ZTE's server, 1.0 version. If it is a helix server, there will be the identity of the Helix server. x

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

type of communication between terminal devices, such as a video session, a time-consuming information processing, or a collaboration session. The agreement does not define or limit the services that can be used, such as transmission, quality of service, billing, security, and other issues that are handled by the underlying core network and other protocols. (1) Contact: SIP and RTSP are Application Layer Control Protocol, responsible for the establishment and control of a communication process

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

.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

big. If you have any questions, I 'd like to ask.I use H. 264 encoder. After reading output_example, I found that the encoded data format should be bound to the avpacket format. I would like to ask, what data format does ffserver use for RTP packaging, does it directly package the cached data after encoding, or does it package the avpacket data using RTP?I hope my eldest brother can give some advice. Reply

eliminate echo interference that has a great impact on the quality of calls and ensure the quality of calls. This is particularly important in IP group networks with relatively large latency. 2.3 real-time transmission technology mainly uses the real-time transmission protocol RTP. RTP is an end-to-end protocol for real-time data transmission, including audio. RTP