iam trying to test my application to public network so i will use my own ip as server , i already did port forward in my router settings, " my goal is reach this application to computers out side my network " but i only can receive data i cannot send from my main ip to other computer ,, i have disabled my firewall also the other pc firewall is this problem of my router ? or my network ?
i tried to disable router firewall but still same problem is there other things i have to check ? also i set my indtCP server binding to 0.0.0.0 to reach all network.
You should be able to send, but only after the client connects to your server.
When a client connects you should handle OnConnect event of your TIdTCPServer component. AThread parameter represent your client. You can respond to it by, for example:
AThread.Connection.Write ('Hi!');
You can disconnect the client by:
AThread.Connection.Disconnect;
Of course, many clients can be connected at the same time. In this case, you should keep track on all (dis)connected clients.
Related
I have installed control room and client successfully and also able to work on it. But, whenever I connect VPN, the control room disconnects automatically with the message "The requested address could not be resolved".
It seems that the control room binds with the system IP, and when connected over VPN system IP changes. As per my understanding, this could be the reason for the disconnection of Control Room over VPN.
Do you have any idea how to resolve this?
Please let me know if you have a solution for this or let me know where can I get help with this issue
yes, this is correct. and this is not an error.
Once the machine is connected to the VPN, then it is effectively on the remote network... so all local resources (including the control room) will not be reachable. most likely you are connecting to the control room using its FQDN.
so you can do one of these things:
while connecting the AA client to the control room, use its IP address instead of hostname
use split DNS on ur VPN or split tunneling
I want to write an app, that establishes direct TCP\IP connection between two devices over the internet.
I have a server to exchange IP addresses and ports, but is it enough to establish a connection?
How to handle cases, where both devices are on the one IP (connected to the same Wi-Fi hotspot)?
Also, I don't understand how open ports on the device could be accessible without port forwarding settings on hotspot? Is there any range of ports open for apps usage?
You are correct about port forwarding for most cases.
TCP/IP is OSI Layer 4 protocol. You need to be able to send data to the sever application with a known IP address and port number. Now when on the internet, this can get more complicated because you normally do not directly interface with the IP address that the rest of the internet reaches you at. This is your public IP address. The modem you have from your ISP likely implements a NAT
, which turns your private address into your public address. You modem will block all TCP SYN requests unless there is forwarding rule specifically for it.
This is just the typical case, you can also look into setting your modem up for DMZ mode and even implementing your own reverse proxy like nginx.
Looking to place calls using our Cisco SPA504G IP phones through Twilio. We have 4 phone lines/numbers with Twilio and we want to use them to place and receive calls on physical phones.
Edited Question:
I found an interesting post in here: https://ertw.com/blog/2013/11/05/using-an-ip-phone-with-twilio/
This where my steps in order to get the phone to ring but I could not hear voice :/
I just bought: http://www.amazon.com/Grandstream-GXP1620-Medium-Business-Device/dp/B00VUU8EZM
Connected phone to my router. I am port forwarding all traffic in the ranges from 10000-20000 to the phone. I am also port forwarding port 5060 to the phone.
Uploaded the following xml file:
<Response>
<Say>Testing</Say>
<Dial>
<Sip>
sip:line1#24.51.221.98
</Sip>
</Dial>
</Response>
that can be found at: http://antnam.com/voip.xml
I called my internet provider and now I have a static IP address so that 24.51.221.98 never changes.
I configured my twilio number (855) 804-0420 to execute a GET # http://antnam.com/voip.xml
When I call (855) 804-0420 I can hear the phone that is connected to my router (voip phone) ring!!! So good news I am able to call the phone I purchased on step 1!
Once I answer the call I am not able to listen to voice :/ . What could I be doing wrong?
In summary everything works great I am just not able to listen to anything. It is as if the call is on mute. Am I missing to open more ports?
It sounds like this may just be a NAT traversal problem; if it is please move or remove this question, as this would only be relevant if you were programming this client. There is a ton of info out there about this issue (for example here is an excellent article that comes up as the first result when I google "voip nat traversal"), but here's a quick summary:
Why NAT causes a problem for VoIP
Most VoIP protocols use a data stream on one port (e.g. 5060 in this case) to negotiate connection information that includes among other things a socket (IP address and port) to receive audio/RTP traffic; there are 2 things about this negotiation of a socket that might be unexpected:
It can be any IP address and port combination, not just one that is on the VoIP device itself. So you might have for instance a VoIP server that negotiates a socket on another host that is not part of the SIP dialog, and which might be behind a NAT
The negotation is done at the OSI Application Layer (Layer 7), so it is normally untouched by the NAT process, which operates at Layers 3 and 4
How to diagnose missing audio due to NAT
If you're able to get packet captures (ideally on both WAN and LAN ports, so you can see your VoIP device's traffic before and after NAT), you can see the problem in action: just look for the packets containing SDP payloads (e.g. if you're doing SIP on UDP 5060, just filter for that port and you will see INVITE requests and 200 OK responses that contain SDP payloads); drill down to the c (Connection Information) and m (Media Description) lines, which should look something like the following:
c=IN IP4 192.168.1.114
m=audio 6094 RTP/AVP 0 8 101
If you're seeing something like this going out your WAN port, it means your VoIP device is requesting to be sent audio on 192.168.1.114:6094; the IP address is a private address and cannot be routed over the internet; the port is just one I picked randomly, but the one you see needs to be open and forwarded to your device
How to fix it
There are various solutions to this problem, which all come down to rewriting the private IP address that your device is giving out into the public one that your device's traffic is being NAT'd out on, so that when the remote device parses the Connection Information line in the SDP, it has a valid, publicly routable IP address to send the audio traffic to, and a UDP port that is NAT'd to your device. Sometimes the VoIP device itself can handle the rewriting (e.g. you can either statically tell the device in its configuration what its public IP is, or it can discover it from a protocol like STUN), sometimes the rewriting is done by the firewall/router that is doing the NAT (there are various names for this, like SIP ALG or SIP Fixup).
Unfortunately due to the variations in NAT implementations across various routers and firewalls, no solution can be guaranteed to work 100% of the time; and if you have multiple devices behind the same firewall, having it do the rewriting will only work for one of them.
In your case:
You mention 2 different VoIP devices, a Cisco SPA504G, and a Grandstream GXP1620. The datasheets for both devices say they support STUN, so I'd start looking in its config for the STUN settings or anything else that references NAT traversal. Also, make sure that the ports you are forwarding to the device are the ones that it uses, this is usually just another item in the config, called something like "RTP base port" or "RTP range"
I would also ensure that you nat transveral enabled with stun using a public stun server such as stun.sipgate.net
Note: STUN operates on TCP and UDP port 3478.
This is required as the phone needs to send the external ip in the sip packets. Without stun it will send the internal ip and the far end sip device will attempt to send the data there.
So I have been trying to create two GCDAsyncSocket's in my project, one (socket) that uploads a file to my server and the other (listenSocket) waits for another process from the server to communicate with it. In my ViewController I have initialized them in viewDidLoad method and setup delegate to self for both sockets.
socket = [[GCDAsyncSocket alloc] initWithDelegate:self delegateQueue:dispatch_get_main_queue()];
listenSocket = [[GCDAsyncSocket alloc] initWithDelegate:self delegateQueue:dispatch_get_main_queue()];
I then made the listenSocket start listening by
NSError *err = nil;
if (![listenSocket acceptOnPort:19920 error:&err]) {
NSLog(#"listenSocket failed to accept: %#", err);
}
I then made socket to connect to remote server and start uploading files.
The problem is that socket works fine and can upload and also read response from my server, but it seems that I can't access the accepting listenSocket by any way. Not by the other process on server, nor by using telnet or by typing the ip address and port number into browser. Why is this and how do I fix it?
Edit:
Here's what I am doing with my app:
I am working on an app that programs for Arduino on iPhone. Due to App Store policy the compiling and uploading process has to be on server, so I'm using socket to upload the code to server to get it compiled. In order to upload the compiled binaries to Arduino, I have to run avrdude which fortunately would accept an ip + port address instead of a usb connection as target. avrdude is implemented so that it connects to the address as a client, so I have to open a listening socket on my app.
I imagine your issue relates to your device not having a routable IP address from the outside world. I have assumed here that you are not testing on a local network with your server and phone both using this network via wifi/cable.
When you are on your device using a mobile network, it is assigned an IP address from the mobile operator. This address is more than likely an address which is part of their internal mobile network. When you connect outside of the mobile network to a server, the address the server sees for your device is not the address you see on the device. The addresses are mapped in transit as your IP packet passes through the various gateways while on its way to the server. Thus when you server sees a connection requested on a listen socket, it has a reply address which when used allows the reply to traverse back to your device.
A similar issue occurs when your device is on WiFi behind a NAT router. Connections made outgoing are seen by the router and it changes the from IP address to be that of the router. Since it sees the start of the conversation, it know where return packets with a given port and sequencing should be routed. However if somebody wants to connect to your device from outside, you have to have set up port forwarding on the router for a known port telling it where to send connection packets.
So applying this to your situation:
Outgoing Works (Why):
Your outgoing socket works, because you are connecting to an externally visible IP. When you connect, the network knows exactly where the packet has to go and the reply address is provided in the packet by the network.
Incoming Does Not Work (Why):
Your listen socket will not be working because the address you are sending to does not exist on the open internet. To make a connection from the server or anywhere else to your device you would need an IP which has a routing mapped through to your device. If the device is on a mobile network, you need an external IP for the mobile network which maps to your device. If the device was behind a NAT router, you would need port forwarding set up.
Solution:
Unfortunately, there is no easy solution as your need an IP address for your device in the outside world. Much depends on your use case which you have not mentioned. You either need an external IP which is reliable or you need to use an intermediate server to handle messaging or you need to change your approach and have the device poll every so often for information.
This is a problem which has existed for a long time and is why peer to peer companies have smart algoithms for connecting peer to peer services which use clever techniques like hole punching to connect devices.
IMHO I would move to a model where your device always initiates the connection if you can.
I am doing socket programming in Delphi 6 Pro using the ICS (TWSocket) library. I know my question may seem either convoluted or awkward, but let me explain my application needs so you understand why I want to do something that goes against the usual convention used with a listening socket, that of spinning off an incoming connection to a new socket returned by the Accept method and continue to listen for new connections on the currently set port.
In my application I accept connections from Skype for sending and receiving audio buffers involved with an active Skype call. The problem is that when Skype connects, there is no handshaking, identification, or authentication that would allow me to know what CALL ID the connection is for. Since Skype can conference calls together, there can be more than one active call at a time. However, I need to know which socket connections belong to which CALL ID.
Since the connection is a "blind" connection as stated above, the only way I can reliably map Skype socket connections to CALL IDs is by controlling carefully the port number I listen on. Then, when I tell Skype to connect the audio for a given CALL ID to a specific port number, I know that a connection coming in on that socket belongs to that Skype CALL ID. For example:
Find an available port number, iPortNumber.
Set my socket to listen on iPortNumber
Tell Skype to connect CALL ID iCallID to PORT number iPortNumber
When I get the SessionAvailable event, I know the incoming Skype connection is for CALL ID iCallID.
Rinse and repeat for each CALL ID I need to handle. I know this means I could end up chewing up a few extra ports but since the number of simultaneous Skype calls is always small I don't see that as a problem. What I am having difficulty with is the standard convention of having a Listening socket that spins off a new socket using Accept when a new connection comes in.
I want to Accept the connection on the Listening socket (same socket), and then specifically stop listening without having to close the connection since I don't want to accept any new connections on that port number anymore. Is there a way to do this?
An alternative avenue would be to use the newly created socket returned by Accept and then close the Listening socket, but then I have to come up with a more complicated method to track port numbers to Skype CALL IDs because, if I am correct in my knowledge, the newly created socket returned by Accept is connected on a different port number than the Listening socket so the Listening socket can keep listening on the existing port number. I'd like to avoid that extra complexity and hassle if I could.
If anyone has a better overall idea/paradigm on how to map my blind Skype connections to the Skype CALL IDs that are tied to them, please let me know. Also, I don't think it's possible, but if there's a clever way to get the process ID behind an incoming Socket connection from a process connecting on the same system as my app, I'd like to know.
One-time listening sockets are not that unusual. The FTP protocol uses them, for instance. Simply create a new listening socket on the desired port (or let the socket decide its own port that you can then retrieve), set its backlog to 1, then call accept() on it just once and close it. If accept() accepts a client connection, it returns a new socket handle that you use to communicate with that client. You don't need to keep the listening socket alive during that time.
I don't know what the ICS equivilent of that operation would be, but in Indy there is a TIdSimpleServer component for exactly this purpose (incidentally, Skype on Windows uses Indy).