Hello!
         i will like to ask that will anyone of you help me understand the Call flows in SIP and also that how do we implement this SIP??
Actually this is my Final year project and i am left with 4 months only....i have studied the related details....

My main work is that when a call is established then i change the IP of the called party i.e. i have two interfaces at the called party end and during the call i make one interface down so my call shifts to another interface or link without the disconnection or call drop you may say???
in need of early reply...

REGARDS
Khattak

when the SAU server (A) send an invite message to CommuniGate PBX (B) , the call occured with nothingg heared from the PBX IVR system, after checking the Sip call flow on the trace we found that the last Ack after the 200 ok message sent to the "TO" field instead of "CONTACT" field as appeared which is not the same as that in RFC 3261, how can we solve this issue and force the CommuniGate to accept the ACK message .

thanx

when the SAU server (A) send an invite message to CommuniGate PBX (B) , the call occured with nothingg heared from the PBX IVR system, after checking the Sip call flow on the trace we found that the last Ack after the 200 ok message sent to the "TO" field instead of "CONTACT" field as appeared which is not the same as that in RFC 3261, how can we solve this issue and force the CommuniGate to accept the ACK message .

thanx

Somebody use VPN to solve the VoIP Blocking issue. But it seems not a good way to solve the voip blocking issue. Because VPN will take more bandwidth and will take effection on the Voice Quality

Currently I am using the VGCP, a new solution to solve the VoIP Blocking issue. Following is theirs website:
http://www.speed-voip.com/index-36.html

If any of you have interested, you may try to use it to solve your VoIP Blocking problems. Thanks.

Andy
andywong-01@hotmail.com

When we make a SIP user by making entry in sip.conf i do not have the option for enable or disable the call waiting ; call waiting is enabled by default There must be this Option to enable of disable, please provide this option soon.


Thanks

Hi,

We provide Voice & Video Softphone

Features
 a.. Customized Skin Interfaces
 b.. Call Timer * Display Balance
 c.. Last Number Redial
 d.. Local Signalization (Dial tone, busy, ring back, etc.) for user
comfort
 e.. Mute
 f.. Touch-Tones
 g.. Address Book
 h.. Microphone Volume Control
 i.. Speaker Volume Control

Benefits

 a.. Use Microsoft g723.1 codec
 b.. Small Application
 c.. Works with any Full-Duplex sound card
 d.. USB headset support.
 e.. Works well on most of the versions of Microsoft Windows ( NT4, 2000,
XP, 2003)
 f.. NAT/Firewall support, stable SIP, RTP ports
 g.. Specify NAT IP to be written in SIP messages
 h.. Auto-Configuration of settings for easy deployment
 i.. STUN support for NAT detection and classification
 j.. Configuration Wizard
 k.. Uses New RFC 3261 compliant stack

If you have any queries, please let me know. I would be happy to assist you. voip200607@yahoo.com

Sir, i post ur url....

Globalization, big corporations rushing overseas to cash in on cheap labor and leaving a lot us high tech workers with low paying jobs or no jobs at all. Seemingly there no end in site. And Americans, being the winners that we are, is winning the race to the bottom.

Well guess what, what goes around comes around and small pockets of new technology out there is biting them in the butt.

Take VOIP for instance, China is shipping some amazing technology over here allowing us to make unlimited long distance calls all over the world at virtually no cost at all. Sorry about that AT&T, and all of the other in and out of state long distance carriers thats been taking us to the cleaners every month. Yes I'm gloating but I've been out of work for most of the last two years with time to ferret out some of those technology pockets and take advantage of their cost savings. IBM cashed in their chips when they sold their PC division to China. I'm just following their lead. I suspect HP is somewhere in that line also.

There's a lot of companies out there selling VOIP hardware. One of many I'm sure, thats selling solid VOIP products at a reasonable price is GrandStream Networks. I've been testing their 'BudgeTone 101' SIP phone, 'HandyTone 286' ATA and their new 'HandyTone 488' on Asterisk (A free PBX software package running an old PC I had laying around).

A 'HandyTone 286', 'BugeTone 101' phone and the PBX is at my house in North Carolina connected to a cable broadband connection. A 'HandyTone 488' and another 'BudgeTone 101' is in Florida at my brothers house connected to a DSL Light broadband connection. Both behind NAT Firewalls. And they work great.

The sound quality is so good you have to try VOIP to believe it. Reliability is also good. Theres no longer waiting to use the phone either. With VOIP you have as many phone lines as you have VOIP phones. And of course theres talking for hours and not worrying about long distance charges even to friends out of state because we can share our local phone lines through the Internet.

It sounds to good to be true right! Three months of testing has convinced me theres very little downside unless you work for a baby Bell or long distance carrier. Those guys are going to be as obsolete as a horse an buggy in a few years.

Here's what we found. Both the good and the Bad.

CONFIGURATION
North Carolina Installation:
Linux Fedora Core 3 (free software)
Asterisk - the latest CVS version available (free software)
Gastman - (A GUI monitor for Asterisk, Also free)
Wildcard X100P - $40 card in my PBX server to access POTS (Plain Old Telephone System)
Handytone 286 - SIP box to connect my regular phones, Also called ATA, (Firmware 1.0.5.16)
Budgetone 101 - SIP phone (Firmware 1.0.5.16)

Florida Installation:
Handytone 488 - access the POTS line and regular phones. (Firmware 1.0.1.0)
Budgetone 101 - SIP phone so we can talk without tying up the regular phones (Firmware 1.0.5.16)

Configuring Asterisk to run on Fedora Core 3 was fairly easy. It took some time to get “music on hold” until I discovered it was as simple as typing “make mpg123”. www.voip.org was a big help there and many other places also. Make them your first stop when configuring Asterisk. Never got Festival (a text to speech function in Asterisk) to work. I ended up writing a shell script instead. Man this stuff is flexible.

You can't beat Grandstream products for the price. At least this week anyway. Their VOIP boxes are easy to configure and work as advertised. Their web interface is not only intuitive but nearly the same on all devices. Once you learn how to configure one , you know how on all their other devices. Phones and ATA's (Analog Telephone Adapter)

Nothing is perfect though so I've listed a couple things you may or may not be interested in.
1.The 'BugeTone 101' SIP phone.
- Two buttons don't don't work. The Message and the conference button. We upgraded the firmware to (1.0.5.22 beta) and got the Message button to work but then a hole can of worms opened up. Most notably NAT (Network Address Translation) transversal went into the proverbial hand basket. It would intermittently go into la-la land also. So back to Firmware 1.0.5.16 we went. Who needs a message button anyway.
The speaker phone is almost unusable due to the absence of echo cancellation circuitry. If two phones are on speaker, the feedback sounds like an old Lost In Space episode.
No indicator for the mute button. I know, picky picky picky, but when talking with someone the sound quality is so good you can't tell if the mute button is on or off. It's not a issue if you leave the mute button alone. Play with it though and you may find yourself mimicking the Verison commercial “Can U Here Me Now?”.

2.The HandyTone 488
- To be fair this is a brand new product. It's amazing it works as well as it does. And the only thing we could find flaky was the call waiting feature. If your talking to someone on the FXS side and another call rings in, you instantly have a one sided conversation. I can hear you but you can't here me. Configure it without call waiting and it's rock solid.

3.Configuration Reliability
VOIP/SIP by it's nature has some built in delay. It digitizes small blocks of your conversation, compresses it, sends it to a PBX in packet form which intern converts it from one format to another if needed, then sends the packets on to their destination which decompress and converts it back into voice. There's more to it than that I know, but thats basically what happens. In extreme cases the packets may have to transverse a number of routers out on the network. In our test environment the shortest route crossed twenty three routers and theres no guarantee packets will take the shortest route. So, in short sometimes we experience delayed conversation causing us to talk on top of each other. We experienced clipped words, which means some of the packets couldn't find their destination. And their were times when it wouldn't work at all due to network outages.

Conclusion:
VOIP over the Internet as it stands now, is plenty reliable enough for personal use. It can virtually eliminate long distance charges on you home and cell phones while also providing a hole array of free features (voice mail, call waiting, callerID, etc..) which your local phone company charges extra for. It's also great for family members that are spread all over the country like mine. No more in state charges when you call a friend out of your calling area when he's only four or five miles down the road. And biggest of all, you don't have to deal with a local phone company that has a monopoly on your area and has laid off their experienced workers, in the name of profits, to the point where residential service is almost non existent. Do I Hear A Hallelujah?

For businesses, VOIP is a great way to manage operating costs in the office giving you economical, reliable, scalable, non pro priority phone system options. The PBX and phones need to stay in the building though. To put it another way, unless you like living on the edge, don't try to run a business that depends on reliable phone service over the Internet. Some VOIP providers out there are selling that very concept to small businesses with disastrous results. The Internet is just not stable enough. Case in point, my Packet8(commercial Internet Phone Provider) phone which I'm also testing has been working fine for the past couple weeks except for a momentary outage now and then. While writing this document, it decided to develop a new feature. I can dial out, but no one can dial in. Packet8 has good phone service and I'm sure they'll fix it as soon as the problem is brought to their attention. The real problem though, is it will be a thirty minute to an hour distraction tomorrow that I don't need. You may say, it sounds like Packet8's problem not the Internet. You may be right. My point is, take Packet8's infrequent glitches compounded with the Internet provider's infrequent glitches (EarthLink) compounded with the broadband access provider's (Time Warner) infrequent glitches compounded with the Internet backbone's (AT&T, Quest, MCI, etc...) infrequent glitches and vol la, you have a business customers can't communicate with from time to time. So do a comprehensive risk assessment before making the leap.

VOIP hardware is not as mature as it could be, the Internet is by no means as stable or reliable as your old PSTN (Public Switched Telephone Network). But to be fair, a stable Internet for VOIP business phone service is coming in the not to distance future. Overwhelming cost savings and flexibility is the motivation.

Its pretty incredible though how much technology they've managed to pack into those little boxes. And theres no doubt that VOIP is going to be the communication medium of choice. Dare I say the next technology revolution.

Here's some excerpts from my asterisk config files:

SIP.CONF
gshtfxs1234 ;HandyTone 488 FXS side
type=friend
callerid=”Florida House FXS”
host=dynamic
nat=yes
qualify=yes
canreinvite=no
dtmfmode=rfc2833 ;set rfc2833 in the GrandStream boxes also
mailbox=1234@default
username=gshtfxs1234
secret=password
disallow=all
allow=ilbc

gshtfxo1234 ;HandyTone 488 POTS side
type=friend
context=gshtfxs1234
callerid=”Florida House FXO”
host=dynamic
nat=yes
qualify=yes
canreinvite=no
dtmfmode=rfc2833 ;set rfc2833 in the GrandStream boxes also
mailbox=1234@default
username=gshtfxo1234
secret=password
disallow=all
allow=ilbc

extensions.conf;
gshtfxs1234
exten => s,1,Playback(transfer,skip) ; Let them know what's going on
exten => s,2,Macro(stdexten,1234,SIP/gshtfxs1234)
exten => s,3,Hangup
;
default
;GSHT-488 to Florida's Pots line
exten => _99,1,Dial(SIP/gshtfxo1234)
exten => _99,2,Congestion

I tried to register, but it wouldn't let me. It kept saying my perfectly valid email address (I tried several) was invalid. So anyway, perhaps you can add a link to my SIP WIki: http://www.toyz.org/cgi-bin/sipwiki.cgi