sip proxy

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Use FireBreath to develop real-time playback interfaces (Yate + SIP + FFMPEG + SDL) and firebreathyate

Use FireBreath to develop real-time playback interfaces (Yate + SIP + FFMPEG + SDL) and firebreathyate At that time, such a blog post was really needed to guide this function module. Unfortunately, FireBreath has very little information on the Internet and is not very familiar with C ++, so we tried and explored it all the way. Fortunately, we have implemented this module, and now we have recorded it. First of all, our Yate

Run the SIP show peers command to view the extension, Agent-Getting Started tutorial series 001.

First, let's take a look at my sip. conf file, as shown below: [201] User= 201 Type = friend Secret = 1234 Host = dynamic Context = incoming [202] User= 202 Type = friend Secret = 1234 Host = dynamic Context = incoming [1234] User= 1234 Type = friend Secret = 1234 Host = dynamic Context = incoming [0] User = 0 Type = friend Secret = 1234 Host = dynamic Context = incoming [1] User = 1 Type = friend Secret = 1234 Host

[Rfc3261] Sip-via Header

In many cases, the SIP does not go directly to the target host, but goes through many intermediate node servers. In the request message, the Via header field indicates the nodes that have passed through (each node passes through, add a via header). In the response message, the via header field indicates the node that the message will go through next (each time the request is returned from the original path, a via header is deleted from each node ). T

Meaning and use of prack in SIP messages

Prack English translation (the provisional Response acknowledgement), you can call IT security information! This compares the image.The final response in the SIP is understood to be reliably transmitted, such as a 200OK response to the invite, and UAC will give an ACK telling UAS that it has received 200OK. The reliability between 200 and ACK is end-to-end. Prack is a mechanism for guaranteeing the reliable transmission of temporary messages (101-199)

Pocket PC hidden sip

In PPC development, you sometimes need to hide the sip. There are many ways to hide the SIP in Windows Mobile 5.0. The following are several methods: 1. shsippreference (m_hwnd, sip_down ); 2. sipinfo Si; Memset ( Si, sizeof (SI )); Shsipinfo (spi_getsipinfo, 0, Si, 0 ); Si. fdwflags = ~ Sipf_on; Shsipinfo (spi_setsipinfo, 0, Si, 0 ); 3. shfullscreen (hdlg, shfs_showtaskbar, shfs_hidesipbutton );

Voice Lab 8-sip Notes

650) this.width=650; "title=" clip_image002 "style=" border-top:0px;border-right:0px;background-image:none; border-bottom:0px;padding-top:0px;padding-left:0px;margin:0px;border-left:0px;padding-right:0px; "border=" 0 "alt = "clip_image002" src= "Http://s3.51cto.com/wyfs02/M00/84/57/wKiom1eNzEnCj0QxAABdCgY5914328.gif" height= "384"/>The difference between H323 and sipSIP P2p:trunkSIP C/S: End pointSIP dialing behavior does not support KPML. Every keystroke is sent once.The default

Mysql-Proxy proxy configuration, mysql-Proxy proxy

Mysql-Proxy proxy configuration, mysql-Proxy proxyI. Principles of Mysql-Proxy Mysql-Proxy is a simple program between your client and Mysql Server. It can monitor, analyze, and change their communication. It is flexible and unrestricted. Its common uses include load balanci

Android Internet telephony development, VOIP/SIP so many open source choose which good

I began to study the VOIP/SIP agreement from 09, open source project also saw a few, the earliest Pjsip 05 began to push the time, began to pay attention to, also in their own winmobile project used. Later also saw Sipdroid,imsdroid (Doubango), Linphone,csipsimple (PJSIP).I think the best advantage of Linphone and Csipsimple,linphone is the full platform support, Android,ios,winphone,windows,linux,mac osx,web all support, but the quality is still unde

Specification for JAIN-SIP Requirement

Requirement Specification Directory 1: Target and scope 1.1jain-Sip API description 2: Introduction 2.1 System Overview 3: requirement definition 3.1 jainsip Target Model 3.2 jainsip naming conventions 3.3 jainsip Structure 3.4 General primitives 4. External requirements 4.1 External Interface 4.2 Resource requirements 4.3 acceptance test release 4.4 documentation requirements 4.5 lightweight requirements 4.6 Quality Requirements 4.7

IOT command (based on sip) client API design for java, iotsip

IOT command (based on sip) client API design for java, iotsipThe Iot Device Control we implement is implemented by extending the sip protocol. Because pjsip is implemented based on pjsip, while pjsip uses C Programming, how to make the business layer (android end, java) easier to use the provided command API is the focus, the original method is to encapsulate (C ---> jni ---> java) from the underlying c, wh

Application of SDP in SIP protocol

streams have a public media format 415 Response ( Media type not supported ) , and join 304 Warning Header field ( Media type not supported ) . 3 . Multicast Operations( 1 ) The multicast addresses that are accepted and sent are the same. ( 2 ) is called not allowed to change the media stream only hair, just accept or receive / To the hair characteristics. ( 3 ) If the call does not support multicasting, the loopback - Response and the Warning ( multicast not available ) . 4 . delayed Media

How to: TCPDUMP sip VoIP capture FreeBSD tutorial

I recently found out doing a sip capture on a FreeBSD system is a little different than centos or other Linux distributions. here is a easy to use command that will grab the SIP packets from TCP dump. this will give you an easy to read text file for debugging or tracing. > Tcpdump-I bce1-n-S0-vvv UDP port 5060>/usr/src/capture_file Let's go over the options for this command: -I = interface which on my BS

Pyhotn's P2P-SIP network phone test

P2p-sip is a peer-to-peer telephone protocol, and someone wrote a python implementation. This only supports python2,2.6 above PIP installation, or download installation package decompression. After decompression has the readme, chews the English. Write webcaller.py Import gevent, sys from gevent import monkey; Monkey.patch_all () from GEVENT.PYWSGI import wsgiserver to CGI import Parse_qs, escape import logging from logging Impor T config logging.co

Important embodiment of SIP Application

The concept of unified communication needs to be understood from the combination of multiple protocols. Among them, there is more participation in the SIP protocol. The integration of unified communication involves software, hardware, mobile devices, and fixed devices. The standards and protocols are different. Therefore, the integration is difficult. For IP phones and desktop applications, the main protocol is S

Discussion on SIP server problems (1)

Some friends may be familiar with the SIP protocol. In this regard, the most prominent thing is the VoIP service. In VoIP services, the SIP protocol and the SIP server are often involved. Next, let's take a look at the traversal problem on the SIP server. 1. Description of SIP

MAC EI Capitan Update system comes with SVN version number (Turn off SIP side sudo rm)

Following last night. decided to update the SVN that comes with the system. The SVN version number that comes with it is 1.7. Crossing Network Svn:http://www.wandisco.com/subversion/download#osx The latest version number is 1.9.13, decided to upgrade under.Unexpectedly due to EI Capitan sip problem Toss a good freshman meeting. Would not have wanted to record. But because sip this egg-ache thing decides sti

Configure a SIP gateway to dial an external phone

If you have a SIP account from a carrier, you can configure the SIP to dial an external phone. The SIP account (or the device providing the account) is referred to as the SIP gateway in FreeSWITCH. Adding a gateway only needs to create an XML file in conf/sip_profiles/external/, with a name that can be randomly used, s

Linux Firewall NAT-SIP network topology, natsip

Linux Firewall NAT-SIP network topology, natsip Environment: Firewall: Ubuntu Server 17.10. SIP-Yate client, Asterisk Server Network Topology: 1) configure the IP address of the firewall and enable forwarding: ifconfig enp2s0 192.168.1.131 ifconfig enp3s0 192.168.100.1 echo 1 > /proc/sys/net/ipv4/ip_forward 2) load the nf_nat_sip module to create expectations: modprobe nf_nat_sip 3) configure th

FreeSWITCH connecting the external SIP Provider

1, added SIP Provider, add the configuration file in Conf/sip_profiles/external This SIP Provider requires REGISTER, and since FreeSWITCH is accessed through NAT, it is set to send pings for 30 seconds. 2, the did mapped by this SIP Provider to the corresponding extension SIP Profile External.xml Sets the context d

WebLogic SIP Server and conference Application

At BEA, my role is to help build and support ISVs of applications on WebLogic Communications Platform to build an ecosystem. It is easy to use WebLogic SIP Server to compile an aggregate J2EE/HTTP/SIP application. This article details the architecture behind a complete conference application that uses Cantata's Media Server to stream audio and video. The two smart developers took less than a month to comp

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