sip presence information

AVAYA's SIP Presence

openSIPS's Presence Server

OpenSer and HA/LoadBalancing

This document mentions Load Balancing about OpenSer.

This also mention this feature.

Here provides a sample configuration file.

# Made by Deillon Thomas
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (cmd line: -E)
children=4
check_via=no
dns=no
rev_dns=no
port=5060

#set module path
mpath="/usr/local/lib64/openser/modules/"

# ------------------ module loading ----------------------------------
loadmodule "maxfwd.so"
loadmodule "sl.so"
loadmodule "dispatcher.so"
loadmodule "tm.so"
loadmodule "mi_fifo.so"
loadmodule "textops.so"
loadmodule "xlog.so"

modparam("mi_fifo","fifo_name", "/tmp/openser_fifo")
#Timer which hits if no final reply for a request
#or ACK for a negative INVITE reply arrives
modparam("tm", "fr_timer", 5)

modparam("dispatcher", "list_file", "dispatcher.list")
modparam("dispatcher", "flags", 2) # fial-over mode
modparam("dispatcher", "dst_avp", "$avp(i:271)")
modparam("dispatcher", "grp_avp", "$avp(i:272)")
modparam("dispatcher", "cnt_avp", "$avp(i:273)")
modparam("dispatcher", "force_dst", 1)

route
{
[...]
if (method=="INVITE") #To complete of course
{
ds_select_dst("1","4"); # 4 = round-robin
t_on_failure("1"); # If there is no response after 5 sec
t_relay();
exit;
}
}

failure_route[1] {
if(t_check_status("408"){ # if timeout
ds_mark_dst(); # do not use this destination anymore
ds_next_dst(); # use next
t_on_failure("1"); # if the next one is dead to !!
t_relay();
}
else {
t_reply("501", "Not Implemented");
}
}

pjsua as a testing client

原文, original

PJSIP: Command-Line VoIP Client for Linux

January 20, 2008 – 11:10 pm

Some time ago, I was tasked at work with finding an IP telephony client that used the SIP protocol, ran on linux, and did everything from the command line. The goal was to have a program that could be deployed at remote networks to test the quality and performance of VoIP calls between the remote site and our central Asterisk VoIP server. I thought this would be an easy task, but it turned out to be far more difficult to find a client that had all of the necessary features. There are a few command-line SIP clients for linux, and most of them are designed for testing, but paradoxically this means that their feature sets have been limited to simply making a call and playing some sort of wav file or reading text from a file. This was handy for sure, but we needed something that would for all intents and purposes be a full-fledged SIP phone that we could control remotely either with a web script or through SSH.

Yesterday I finally stumbled across a program that fit the bill. More than just a program with a static set of features, PJSIP contains a complete library of functions so that you can build your own programs based on the SIP protocol in practically any language you like. However, it does come with several example applications you can build and use on multiple platforms. The primary application is called pjsua, and contains a fully-featured, menu-based interface for the command line. This program alone did everything we needed and more.

Pjsua is incredibly easy to use. Run without any options, it presents you with a nice menu-interface where you can add accounts, add and view the status of other SIP users, make calls, transfer calls, put calls on hold, conference calls, and even pipe in audio from wav files. If you’re in a hurry, you can make a call with the following short and simple command:

pjsua sip:<user>@<domain>

Or if you know the IP address of the other user, you can do the following:

pjsua sip:<ip address>

Chances are you’ll want to have a registered account on a server running something like Asterisk. You can use command-line options to specify the account details every time, or you can put the command-line options into a config file and have pjsua load that instead using the following syntax:

pjsua –config-file=<config file>

One of the neatest features of pjsua is the ability to run it in “daemon” mode, officially called auto-answer. In this mode, pjsua sits idling, waiting for an incoming call. When it detects an incoming call, pjsua answers and can do several different things. Here is an example command:

pjsua –null-audio –play-file=data3.wav –auto-play –rec-file=call001.wav –auto-rec –auto-answer=200 –config-file=pj-config

Broken down, this command tells pjsua to run, register itself with the central server with the details provided in “pj-config”, to listen for incoming calls and automatically answer with code 200 (for those of you familiar with HTTP status codes, SIP status codes are remarkably similar), to loop data3.wav after answering the call, and to record any audio on the line (except the wav file being piped in).

Why on earth would this be useful? Say, for instance, I have a business client using my VoIP service at the other end of the state. He calls me to complain that call quality has been atrocious for the past few days. There are a few things I can do. I can take up a bunch of his time having him troubleshoot with me on the phone, I can drive all the way across the state and troubleshoot the issue myself, or I can use the web server we’ve also installed on his network to make my own test calls without ever leaving my office or tying up the client’s time or phone line. I can simply place a call to the test number and listen to the looped recording for evidence of audio quality issues. This only tells me half the story however, as often times with VoIP sessions only one side of the conversation is affected. This is where the recording comes in, as I can now download it from the remote server and analyze it for problems. This saves both time and money, and provides an easy yet effective way to track down problems.

Pjsua is a fully featured SIP client, so if you have speakers and a microphone on your system, you can use this just like you would any other soft-phone. However, given that there are several other soft-phones for linux with a nice GUI interface, one of them may be a better choice if your goal is simply to make and receive calls from your computer. But if you need something to run in a windowless environment, or you’re like me and need something lightweight to help troubleshoot VoIP issues, you can download Pjsua and the entire PJSIP library from http://www.pjsip.org.

some sip products url

EutelVoIP

Home at PortaOne - VoIP Solutions Company

VaxVoIP SIP SDK includes SIP activeX, SIP Lib, SIP Dll and SIP Cabinet

Signaling gateway, SS7 signaling converter, IP PRI gateway, media gateway, SS7 gateway

FreeSWITCH | Communication Consolidation

IP PBX - 3CX PBX, PABX software for Windows

http://www.3cx.com/

there is a free version.

pjsip

這兩天仔細閱讀了 PJSIP-Dev-Guide.pdf, 真是獲益良多.
document 的結構非常清楚 表達了一個 SIP protocol stack 架構.
並且闡明如何利用 一些簡單的規則 來撰寫程式.
分層負責/管理 的方式, 除了容易理解程式, 也更好使用.

debug my first SIP application

昨天的 capture file 顯示還有一些問題. (但是沒有在現場發現).

原來我把一些 From/To 的原則搞錯了. 並且 ACK/CANCEL 的部份也處理的不太好.
再加上處理 TS 出問題, B-Party/A-Party 在未接通前先掛機的問題也沒有處理.

還好, 對方說今天不能測, 讓我有機會把這些問題一一澄清, 改好.

現在真的只剩 PRACK 而已.

當然, 要做一個真正的 AP, timer 部份還是得好好處理.

my first SIP application

今天到 S 公司去測試 SIP A/S.

很高興, 通了. 比預計花的時間短. S 公司的人也很高興.

這是第一個完全由我撰寫的 SIP 測試工具. 很有成就感.
這個過程也讓我看了更多的相關資料. (關於 3GPP)

SIP/ISC - 3

繼續閱讀, 並翻閱相關的 source.

原來 ser 的 parser 是用到那裡就 parse 到那裡. 而且大量使用 convert string to lower case then treate 4 octets as unsigned long inter to speed up the comparision.

有趣.

SIP/ISC - 2

繼續閱讀
SER - Getting Started 5 - 6.

http://www.tech-invite.com/ - 繼續看這個網站, 真是很棒.
例如: 講到 call flow, 他會有圖示; 講到 SIP 的各個欄位, 會用 ABNF 語法來表示, 還會列出相關的 rfc 出處. 真的是很棒的網站, 應該要 mirror 下來.

SIP/ISC

今天搜尋進一步的相關資料. 記下來免得忘了.

http://www.iptel.org/files/sip_tutorial.pdf 中有一章節, 介紹 SIP 在 IMS 的環境.

http://www.onsip.org/ 是一個不錯的入口網站,
http://www.internet2.edu/sip.edu/

http://www.tech-invite.com/ - a Portal devoted to SIP and surrounding technologies
最大的優點是, 對於相關的資訊都有整理過, 不是光 SIP Protocol 本身, 與 IMS/PSTN 相關得東西都有涉獵.

閱讀:
1. 比較 ser 及 openser 的文章. - (還是 用 ser 好了. 但別忘了 openser 有 Cisco 的產品所採用)
2. SER - Getting Started 1 - 4.