Shulou(Shulou.com)06/01 Report--

RTPProxy

Rtp media agent usage and Win32 migration.

KamailioIMS

Use kamailio to build IMS instances.

Implementation Analysis of Qos in Linphone

1. RTP statistics: packet loss, jitter, delay, bandwidth usage

2 、 RTCP

3 、 TOS / DSCP

4. RSVP resource reservation

5 、 Jitter buffer

6. Adaptive jitter compensation

7. Automatically adjust the transmission bit rate

H264 format

The relationship between H264 Annnex format and RTP network transmission format.

H264 processing source code

H264 common handling:

1. Use Ffmpeg and X264 to encode and decode H264

2. H264 ES RTP transceiver

3. RTP packet of H264 by Linphone

4. RTP transfers to TS file after receiving H264 ES

H264 RTP2Annexb

Extract H264 from RTP message and save it as Annnex file.

Rtp2h364 × ×

Using Asterisk as the gateway for SIP and MGCP protocols

Use Asterisk-13.6.0 as the SIP and MGCP protocol gateways to initiate calls from the SIP and MGCP sides, respectively.

Use Asterisk-13.6.0 as the SIP and MGCP protocol gateways to initiate calls from the SIP and MGCP sides, respectively.

RTSP protocol instance

RTSP (Real Time Streaming Protocol), RFC2326, real-time streaming protocol.

H248 protocol

A comprehensive introduction of the H248/MEGACO protocol.

SIP video terminal SIPUA

Windows SIP video terminal, based on linphone development. Optimize linphone call speed, DNS parsing, routing, simplify configuration, resolve memory leaks and other bug. Supports H264 and H263-1998 video coding, using libraries such as osip, eXosip, oRTP, mediastreamer, ffmpeg (AVCodec, AVDevice, AVFilter, AVFormat, AVUtil). For more information on sipua.cfg configuration, please see sipua.log for the log.

SIPUA download

Asterisk speech processing

Asterisk voice processing has nothing to do with business and is a unified process. The processing objects are RTP and File. Regardless of the input format of external RTP or File, it is converted to signed linear format by default after entering Asterisk, and then from slinear format to external RTP or File format when exporting from Asterisk (sending or writing files).

RTP Voice recovery tool RTP2Wav

RTP voice recovery tool, filter pcap, cap package voice saved as wave format files, support G711, G723, G726, G729, ilbc, gsm, speex, L16/SLINEAR, DVI4/ADPCM format, is a good tool for voip professionals.

Rtp2wav download

Voice format conversion tool convert

Voice media format conversion tool, which supports the conversion of wav, vox, sln, raw, alaw, al, pcm, ulaw, ul, mu, ogg, ilbc, gsm, G729, G726, G723, au, mp3 and other voice formats.

Convert download

Mixing tool mixer

Audio mixing tool, which supports audio mixing in mp3, wav, vox, sln, raw, alaw, al, pcm, ulaw, ul, mu, ogg, ilbc, gsm, G729, G726, G723, au formats. Currently, you can only mix two files.

Mixer download

Using SIPp for SIP stress Test

Use sipp for sip registration, call, conference stress testing.

Implementation of VOIP load balancer

A VOIP Loadbalancer developed on the basis of Asterisk, compatible with SIP,H323 and IAX protocols, 7 load balancing strategies!

In addition to the load balancing function, it also supports routing retry, number conversion and protocol conversion functions.

Using Asterisk as SIP and H323 gateways

Asterisk supports SIP,H323,IAX,MGCP and other VOIP protocols, and can be used as a variety of protocol gateways. This paper introduces how to use Asterisk as a SIP-H323 protocol gateway.

SIP routing

Explain the routing mechanism of SIP request and response through the Request-URI,Via,Record-Route,Route,Contact protocol header.

H323 protocol

A comprehensive introduction of the H323 protocol.

SIP protocol training

SIP Protocol training 1

SIP Protocol training 2

SIP protocol

A comprehensive introduction of the SIP protocol.

SIP platform architecture

Take SER as an example to introduce the SIP Proxy architecture, including media traversing NAT,Radius prepaid implementation.

SER installation

SER basic module installation configuration, including database, NAT traversal.

STUN protocol

The working principle and process of STUN in SIP traversing NAT.

NAT traversing technology

SIP traverses NAT technology, and some are also applicable to H323.

Traversing NAT in Citron mode

Citron NAT traversing method solves the problem of H323 terminal being called under private network.

T38 fax

The technology of transplanting fax to Ethernet for transmission.

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