SIP Stack with SIP Proxy - (VOIP)

SIP Overview

SIP is an application-layer control protocol that can establish, modify, and terminate multimedia sessions (conferences) such as Internet telephony calls. SIP can also invite participants to already existing sessions, such as multicast conferences. Normally SIP uses UDP and TCP port 5060 and TCP 5061 for SSL communication. SIP protocol is very similar to HTTP, so if you have some knowledge about HTTP, then it is easy to learn SIP. SIP doesn't transfer session data like audio, video. RTP(real time protocol) is used for that, SIP just helps to open RTP streams.

SIP Message Example

INVITE sip:john@domain.com SIP/2.0
From: <sip:doe@domain.com>;tag=2084442460
To: <sip:john@1domain.com>
Via: SIP/2.0/UDP domain.com:5060;branch=z9hG4bK2df7b9194cd51e25
Call-ID: john@domain.com-4524j
CSeq: 1 INVITE
Contact: <sip:doe@domain.com:5060>
Content-Length: 226
Content-Type: application/sdp

<session description data, like RTP description>

SIP Server Types

stateless	SIP server doesn't store any transaction info.
statefull	SIP server creates and holds SIP commands transaction state.
registrar/location	Allows users to register their locations and later to use that info to forward calls to registered contact.
B2BUA	SIP server is like statefull + holds active calls state.(This is needed if call billing or full control of call is needed)
presence	Provides user availability services, like if user is online,offline, ... .
...	There are some more, but not so important ones.

Basic SIP Commands

INVITE - Initiates a session. This method includes information about the calling and called users and the type of media that is to be exchanged.
ACK - Sent by the client who sends the INVITE. ACK is sent to confirm that the session is established. Media can then be exchanged.
BYE - Terminates a session. This method can be sent by either user.
CANCEL - Terminates a pending request, such as an outstanding INVITE. After a session is established, a BYE method needs to be used to terminate the session.
OPTIONS - Queries the capabilities of the server or other devices. It can be used to check media capabilities before issuing an INVITE.
REGISTER - Used by a client to login and register its address with a SIP registrar server.

Ok, some ABC done, there are many documents on the internet, so it is not a good idea to rewrite these there.

If want more advanced information, then see:

RFC 3261 (defines SIPv2)
SIP Introduction
A Newcomer's Guide to SIP
A Portal devoted to SIP and surrounding technologies
Google.com is always your friend

SIP Proxy Demo Overview

This SIP proxy example just implements fully functional simple stateless, statefull, b2bua proxy. You can use hardware SIP phones or soft phones to play with this proxy.This is an advanced example, code is well commented, so beginners don't hate me because no more text here. Just read RFC 3216, see information links I noted earlier. After you go through those, if you then look at the code, it is all nicer then.

Some free available softphones are:

Version:
    11.06.2007
        *) Added B2BUA support.
    07.04.2007 
        *) Many bug fixes. 
        *) SIP -> PSTN and PSTN -> SIP gateway support. 
        *) Non-SIP URI gateway support.

Contact Details

Email: ivar@lumisoft.ee
Forum: http://www.lumisoft.ee/Forum

License

This article, along with any associated source code and files, is licensed under The Code Project Open License (CPOL)

About the Author

Ivar Lumi

Estonia

Member

No Biography provided

Article Top

Poor

Excellent

Add a reason or comment to your vote: x
Votes of 3 or less require a comment

Comments and Discussions

You must Sign In to use this message board.

Search this forum
	Profile popups Spacing Noise Layout Per page

study sip in my project
Sign In·View Thread·Permalink

everything working fine but when i try to calling does not work i have a sip server any help please!!
Sign In·View Thread·Permalink

class SDP 329 if(!string.IsNullOrEmpty(this.Originator)){ 330 331 retVal.AppendLine("0=" + this.Originator); 332 333 } modified 22 Jun '12 - 3:23.
Sign In·View Thread·Permalink

169 private void RemoveExpiredNonces() 170 { 171 lock(m_pNonces){ 172 for(int i=0;i if(m_pNonces[i].CreateTime.AddSeconds(m_ExpireTime) > DateTime.Now){ 175 m_pNonces.RemoveAt(i); 176 i--; 177 } 178 } 179 } 180 } 179 } 180 }
Sign In·View Thread·Permalink

Hi Yes it should be "<"
Sign In·View Thread·Permalink

IPPhone IP is 192.168.1.2; Sip Proxy IP is 192.168.1.253
When IPPhone Register , the SIP fails the process and shows the error.
What can I do to resolve it ?
////// error log//////////
System.FormatException: 輸入字串格式不正確。(translation: input string wrong)
於 System.Number.StringToNumber(String str, NumberStyles options, NumberBuffer& number, NumberFormatInfo info, Boolean parseDecimal)
於 System.Number.ParseInt32(String s, NumberStyles style, NumberFormatInfo info)
於 System.Convert.ToInt32(String value)
於 LumiSoft.Net.AUTH.Auth_HttpDigest.Parse(String digestResponse) 於 D:\Csharp Book\LumiSoft.Net\Net\Net\AUTH\Auth_HttpDigest.cs: 行(row) 126
於 LumiSoft.Net.AUTH.Auth_HttpDigest..ctor(String digestResponse, String requestMethod) 於 D:\Csharp Book\LumiSoft.Net\Net\Net\AUTH\Auth_HttpDigest.cs: 行 35
於 LumiSoft.Net.SIP.SIP_Utils.GetCredentials(SIP_Request request, String realm) 於 D:\Csharp Book\LumiSoft.Net\Net\Net\SIP\SIP_Utils.cs: 行 145
於 LumiSoft.Net.SIP.Proxy.SIP_Proxy.AuthenticateRequest(SIP_RequestReceivedEventArgs e, String& userName) 於 D:\Csharp Book\LumiSoft.Net\Net\Net\SIP\Proxy\SIP_Proxy.cs: 行 823
於 LumiSoft.Net.SIP.Proxy.SIP_Proxy.ForwardRequest(Boolean statefull, SIP_RequestReceivedEventArgs e, Boolean addRecordRoute) 於 D:\Csharp Book\LumiSoft.Net\Net\Net\SIP\Proxy\SIP_Proxy.cs: 行 351
於 LumiSoft.Net.SIP.Proxy.SIP_Proxy.OnRequestReceived(SIP_RequestReceivedEventArgs e) 於 D:\Csharp Book\LumiSoft.Net\Net\Net\SIP\Proxy\SIP_Proxy.cs: 行 174
///////reg log////////
Transaction [branch='z9hG4bK4e8808b81af58f0f0f98f9fae0fa8ac6';method='REGISTER';IsServer=true] timer I(Non-INVITE request retransmission wait) triggered.
Transaction [branch='z9hG4bK4e8808b81af58f0f0f98f9fae0fa8ac6';method='REGISTER';IsServer=True] switched to 'Terminated' state.

Request [method='REGISTER'; cseq='781'; transport='UDP'; size='401'; received '192.168.1.2:5060' -> '192.168.1.253:5060'.

REGISTER sip:192.168.1.253 SIP/2.0
Call-ID: 4099112003-3E8B-0007
Contact:
CSeq: 781 REGISTER
Expires: 3600
From: "101" ;tag=50-831083433
Max-Forwards: 70
To: "101"
User-Agent: AmRoad
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;branch=z9hG4bKed119c3b6e431f736168068fd9086387;received=192.168.1.2
Content-Length: 0

Transaction [branch='z9hG4bKed119c3b6e431f736168068fd9086387';method='REGISTER';IsServer=true] created.
Transaction [branch='z9hG4bKed119c3b6e431f736168068fd9086387';method='REGISTER';IsServer=True] switched to 'Trying' state.
Response [flowReuse=true; transactionID='z9hG4bKed119c3b6e431f736168068fd9086387'; method='REGISTER'; cseq='781'; transport='UDP'; size='505'; statusCode='407'; reason='Proxy Authentication Required'; sent '192.168.1.253:5060' -> '192.168.1.2:5060'.

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;branch=z9hG4bKed119c3b6e431f736168068fd9086387;received=192.168.1.2
From: "101" ;tag=50-831083433
To: "101" ;tag=91b64c1f81bef5e95e30cea3
Call-ID: 4099112003-3E8B-0007
CSeq: 781 REGISTER
Allow: INVITE,ACK,CANCEL,BYE,MESSAGE
Proxy-Authenticate: digest realm="",nonce="61bc850c786241f19f57580448a9480b",opaque="1d795f4460ec4cfd9892f4ecd50d91c7"
Content-Length: 0

Transaction [branch='z9hG4bKed119c3b6e431f736168068fd9086387';method='REGISTER';IsServer=True] switched to 'Completed' state.
Transaction [branch='z9hG4bKed119c3b6e431f736168068fd9086387';method='REGISTER';IsServer=true] timer J(Non-INVITE request retransmission wait) started, will trigger after 32000.
Request [method='REGISTER'; cseq='782'; transport='UDP'; size='626'; received '192.168.1.2:5060' -> '192.168.1.253:5060'.

REGISTER sip:192.168.1.253 SIP/2.0
Call-ID: 4099112003-3E8B-0007
Contact:
CSeq: 782 REGISTER
Expires: 3600
From: "101" ;tag=50-831083433
Max-Forwards: 70
Proxy-Authorization: Digest username="101@test.com",realm="",nonce="61bc850c786241f19f57580448a9480b",uri="sip:192.168.1.253",response="5dce688c47b46a36dddfe8b8c162fbf8",opaque="1d795f4460ec4cfd9892f4ecd50d91c7",nc=0000030e
To: "101"
User-Agent: AmRoad
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;branch=z9hG4bK32d8da85d26639c06759a5a1082c56fa;received=192.168.1.2
Content-Length: 0

Response [transactionID=''; method='REGISTER'; cseq='782'; transport='UDP'; size='409'; statusCode='500'; reason='Server Internal Error: 輸入字串格式不正確。'; sent '' -> '192.168.1.2:5060'.

SIP/2.0 500 Server Internal Error: 頛詨摮葡?澆?銝迤蝣箝?
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;branch=z9hG4bK32d8da85d26639c06759a5a1082c56fa;received=192.168.1.2
From: "101" ;tag=50-831083433
To: "101" ;tag=0f7c40ad88cb505d4fb3ece3
Call-ID: 4099112003-3E8B-0007
CSeq: 782 REGISTER
Allow: INVITE,ACK,CANCEL,BYE,MESSAGE
Content-Length: 0

Transaction [branch='z9hG4bKed119c3b6e431f736168068fd9086387';method='REGISTER';IsServer=true] timer I(Non-INVITE request retransmission wait) triggered.
Transaction [branch='z9hG4bKed119c3b6e431f736168068fd9086387';method='REGISTER';IsServer=True] switched to 'Terminated' state.

Hi, One thing what i see ... is : To: Can not be just "101", it must be URI. Normally valu like "sip:101@somedomian.com"
Sign In·View Thread·Permalink

Finally I set the Registrar as URL, than Register process is successful.
IP Phone : IP: 192.168.1.2, UserName: 101, Account: 101, PassWord:1234 , SIP Proxy: 192.168.1.253, Registrar: test.com
SIP Proxy: Setting Tab -> IP: 192.168.1.253, User Account: 101, PassWord:1234 , Address: 101@test.com
///////////////
But new problem happen. After the Communication between two IP Phones and the sip flows disposed, I redial a phone but get a error message as following:
=========error log======================================
System.ObjectDisposedException: Fail to access the Disposed object 。
object name: 'SIP_Flow'。
於 LumiSoft.Net.SIP.Stack.SIP_Flow.get_ID() 於 D:\Csharp Book\LumiSoft.Net\Net\Net\SIP\Stack\SIP_Flow.cs: 行 477
於 LumiSoft.Net.SIP.Proxy.SIP_ProxyContext.TargetHandler.SendToFlow(SIP_Flow flow, SIP_Request request) 於 D:\Csharp Book\LumiSoft.Net\Net\Net\SIP\Proxy\SIP_ProxyContext.cs: 行 615
於 LumiSoft.Net.SIP.Proxy.SIP_ProxyContext.TargetHandler.Start() 於 D:\Csharp Book\LumiSoft.Net\Net\Net\SIP\Proxy\SIP_ProxyContext.cs: 行 532
於 LumiSoft.Net.SIP.Proxy.SIP_ProxyContext.Start() 於 D:\Csharp Book\LumiSoft.Net\Net\Net\SIP\Proxy\SIP_ProxyContext.cs: 行 1025
於 LumiSoft.Net.SIP.Proxy.SIP_Proxy.ForwardRequest(Boolean statefull, SIP_RequestReceivedEventArgs e, Boolean addRecordRoute) 於 D:\Csharp Book\LumiSoft.Net\Net\Net\SIP\Proxy\SIP_Proxy.cs: 行 587
於 LumiSoft.Net.SIP.Proxy.SIP_Proxy.OnRequestReceived(SIP_RequestReceivedEventArgs e) 於 D:\Csharp Book\LumiSoft.Net\Net\Net\SIP\Proxy\SIP_Proxy.cs: 行 174
===========sip log=============================
Transaction [branch='z9hG4bK0b3458bd91bbc9dbac04f955c639d475';method='BYE';IsServer=True] switched to 'Completed' state.
Transaction [branch='z9hG4bK0b3458bd91bbc9dbac04f955c639d475';method='BYE';IsServer=true] timer J(Non-INVITE request retransmission wait) started, will trigger after 32000.
Transaction [branch='z9hG4bK-75d6c012c1be4f818bf8a145592ddeee';method='BYE';IsServer=False] switched to 'Completed' state.
Transaction [branch='z9hG4bK-75d6c012c1be4f818bf8a145592ddeee';method='BYE';IsServer=false] timer K(Non-INVITE 3xx - 6xx response retransmission wait) started, will trigger after 5000.
Transaction [branch='z9hG4bK-75d6c012c1be4f818bf8a145592ddeee';method='BYE';IsServer=false] timer K(Non-INVITE 3xx - 6xx response retransmission wait) triggered.
Transaction [branch='z9hG4bK-75d6c012c1be4f818bf8a145592ddeee';method='BYE';IsServer=False] switched to 'Terminated' state.
Transaction [branch='z9hG4bK-75d6c012c1be4f818bf8a145592ddeee';method='BYE';IsServer=false] disposed.
Transaction [branch='z9hG4bK-75d6c012c1be4f818bf8a145592ddeee';method='BYE';IsServer=False] switched to 'Disposed' state.
ProxyContext(id='c4251e80-8ba0-40bb-b172-97f8980427fa') disposed.
Request [method='INVITE'; cseq='8'; transport='UDP'; size='780'; received '192.168.1.2:5060' -> '192.168.1.253:5060'.

INVITE sip:103@test2.com SIP/2.0
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,REFER,UPDATE
Call-ID: 4190661699-2B8D-0006
Contact: "102"
Content-Type: application/sdp
CSeq: 8 INVITE
Expires: 180
From: "102" ;tag=40-1766746275
Max-Forwards: 70
Supported: replaces,timer
To:
User-Agent: AmRoad
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;branch=z9hG4bK3e67bd1a77d7ca7954d700aceac28d13;received=192.168.1.2
Content-Length: 263

v=0
o=102 4190661697 4190661697 IN IP4 192.168.1.2
s=Session SDP
c=IN IP4 192.168.1.2
t=0 0
m=audio 9016 RTP/AVP 18 0 8 101
a=ptime:20
a=rtpmap:18 G729/8000
a=rtpmap Blush | :O

PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

Transaction [branch='z9hG4bK3e67bd1a77d7ca7954d700aceac28d13';method='INVITE';IsServer=true] created.
Transaction [branch='z9hG4bK3e67bd1a77d7ca7954d700aceac28d13';method='INVITE';IsServer=True] switched to 'Proceeding' state.
ProxyContext(id='5b232fdc-fdae-4828-a96a-e771034269e4') created.
Response [transactionID=''; method='INVITE'; cseq='8'; transport='UDP'; size='423'; statusCode='500'; reason='Server Internal Error: Fail to access the Disposed object。
object name: 'SIP_Flow'。'; sent '' -> '192.168.1.2:5060'.

SIP/2.0 500 Server Internal Error:
object name: 'SIP_Flow'
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;branch=z9hG4bK3e67bd1a77d7ca7954d700aceac28d13;received=192.168.1.2
From: "102" ;tag=40-1766746275
To: ;tag=a18f4a78a3a52b0e8ff37026
Call-ID: 4190661699-2B8D-0006
CSeq: 8 INVITE
Allow: INVITE,ACK,CANCEL,BYE,MESSAGE
Content-Length: 0

Request [transactionID='z9hG4bK3e67bd1a77d7ca7954d700aceac28d13'; method='ACK'; cseq='8'; transport='UDP'; size='332'; received '192.168.1.253:5060' <- '192.168.1.2:5060'.

ACK sip:103@test2.com SIP/2.0
Call-ID: 4190661699-2B8D-0006
CSeq: 8 ACK
From: "102" ;tag=40-1766746275
Max-Forwards: 70
To: ;tag=a18f4a78a3a52b0e8ff37026
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;branch=z9hG4bK3e67bd1a77d7ca7954d700aceac28d13;received=192.168.1.2
Content-Length: 0

Response [flowReuse=true; transactionID='z9hG4bK3e67bd1a77d7ca7954d700aceac28d13'; method='INVITE'; cseq='8'; transport='UDP'; size='344'; statusCode='100'; reason='Trying'; sent '192.168.1.253:5060' -> '192.168.1.2:5060'.

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;branch=z9hG4bK3e67bd1a77d7ca7954d700aceac28d13;received=192.168.1.2
From: "102" ;tag=40-1766746275
To: ;tag=a18f4a78a3a52b0e8ff37026
Call-ID: 4190661699-2B8D-0006
CSeq: 8 INVITE
Allow: INVITE,ACK,CANCEL,BYE,MESSAGE
Content-Length: 0

Transaction [branch='z9hG4bK0b3458bd91bbc9dbac04f955c639d475';method='BYE';IsServer=true] timer I(Non-INVITE request retransmission wait) triggered.
Transaction [branch='z9hG4bK0b3458bd91bbc9dbac04f955c639d475';method='BYE';IsServer=True] switched to 'Terminated' state.

Conatct and from fields must be URI too, not sure if that changes anything.
Sign In·View Thread·Permalink

THX ! But My IP Phone(wvp-800) do not have a item to set domain name but IP. ps: IP Phone : model: wvp-800 url: http://www.welltech.com/product_e_0l.htm
Sign In·View Thread·Permalink