I have a SIP Server running OpenSIPs 1.11.3
configured with built-in STUN module (full mode with 2 IPs)
configured with MediaProxy 2.6.1 to relay RTP (using engage_media_proxy in routing script)
Using IMSDroid from doubango as the SIP client.
Calls between wifi-wifi is good, I do not need to turn on any STUN, ICE, TURN option in the client.
However, calls between 3g-wifi or 3g-3g isn't that good. 3G can make outgoing call but it cannot receive call. Which means 3g-3g call can NEVER happen. All I see in the OpenSIPs logs are repeated retransmissions of INVITE because it cannot reach the 3G side.
I read that TURN server can solve this kind of problem, so I enabled TURN in IMSDroid sip client, but still 3G side cannot receive any call.
The TURN server I am using:
url: 'turn:numb.viagenie.ca'
credential: 'muazkh'
username: 'webrtc#live.com'
Is there any solution / module I can use to solve this problem?
EDIT:
If I use TCP protocol, I am able to receive call! Although the call terminate due to transport error after 30 seconds, but at least the call went through. Any idea what happen here?? Mobile carrier blocking incoming call? But definitely not port blocking because I am able to register whether I use port 80 or 5060.
EDIT 2:
I tried using free SIP accounts to make calls (sip2sip.info and sip.antisip.com), and I have the same problem too! As I know, sip2sip.info is using OpenSIPS too but AntiSip.com is using something like AmSIP. So the problem is with my mobile carrier?
Thank you!
If your UA can't receive calls, it means it is not reachable for signaling. In order for your UA to be reachable, it needs to register and keep the NAT mappings alive. To keep a NAT mapping alive, your UA must send keepalive to the server periodically. Another option is that the server sends keepalives to the UA but some NATs don't refresh mappings for incoming traffic.
When you solve this first issue, comes the media part where technologies like STUN, TURN and ICE will help.
Related
I am making video calls from Linphone to linphone on android.Both the mobile apps are successfully registered on Freeswitch server.
But when i make a call it does't get established.It is happening randomly only.When i checked dump on server i found that Freeswitch keeps sending Invite on B leg side but does't get any response for invite.There are also multiple sip uri's in invite.
Can somebody help me with this?
Is there anything related to server configuration?
Thanks
It's frequent, call over UDP doesn't work.
Try with TCP and normally it's must work correctly.
For more information about the difference between TCP and UDP look at:
https://stackoverflow.com/a/5970545/7131120
I have a client and a server application that is communicating just fine, there is a TIdCmdTCPServer in the server and a TIdTCPClient in the client.
The client has to authenticate in the server, the client asks the server for the newest version information and downloads any updates, and other communications. All this communication with TIdTCPClient.SendCmd() and TIdTCPClient.LastCmdResult.Text.Text.
The way it is, the server receives commands and replies, the clients only receives replies, never commands, and I would like to implement a way to make the client receives commands. But as I heard, if the client uses SendCmd it should never be listening for data like ReadLn() as it would interfere with the reply expected in SendCmd.
I thought of making a command to check for commands, for example, the client would send a command like "IsThereCommandForMe" and the server would have a pool of commands to each client and when the client asks, the server send it in the reply, but I think it would not be a good approach as there would be a big delay between the commands being available and the client asking for it. I also thought of making a new connection with new components, for example a TIdCmdTcpClient, but then there would be 2 connections for each client, I don't like that idea as I think it could easily give problems in the communication.
The reason I want this, is that I want to implement a chat functionality in the client, and it should be receiving messages from the server without asking for it all the time, imagine all clients continually asking the server if there is message for them. And I would like to be able to inform the client when there is an update available instead the client being asking if there is any. And with this I could send more commands to the client too.
what are your thoughts about this ? how can I make the server receiving commands from the clients, but also sends them ?
TCP sockets are bidirectional by design. Once the connection between 'client' and 'server' has been established, they are symmetric and data can be sent at any time from any side over the same socket.
It only depends on the protocol (which is just written 'contract' for the communication) which communication model is used. HTTP for example uses a request/reply model. With Telnet for example, both sides can initate data transmissions. (If you take a look at the Indy implementation for Telnet, you will see that it uses a background thread to listen for server data, but it uses the same socket connection in the main thread to send data from client to server).
A "full duplex" protocol which supports both request/response and server push, and also is firewall-friendly, is WebSockets. With WebSockets (a HTTP upgrade), the server can send data to the connected client(s) any time. This would meet your 'chat' requirement.
If you use TIdTCPClient / TIdCmdTCPServer, corporate firewalls might block the communication.
I'm trying to spec out the foundations for a server application who's purpose will be to..
1 'receive' tcp and/or udp packets
2 interpret the contents (i.e. header values)
To add more detail, this server will receive 'sip invites' and respond with a '302 redirect'.
I have experience with Net::Pcap and perl, and know I could achieve this by looping for filtered packets, decoding and then using something like Net::SIP to respond.
However, there's a lot of bloat in both of these modules/applications I don't need. The server will be under heavy load, and if I run TCPDUMP on it's own, it loses packets in the kernel due to server load, so worry it wont be appropriate :(
Should I be able to achieve the same thing by 'listening' on a socket (using IO::Socket for example) and decoding a packet?
Unfortunatly by debugging, it's hard to tell if IO::Socket will give me the opportunity to see a raw packet? And instead it automatically decodes the message to a readable format!
tl;dr: I want to capture lots of SIP Invites, analyse the head values, and respond with a SIP 302 redirect. Is there a better way than using tcpdump (via Net::Pcap) to achieve this?
Thanks,
Moose
Is there a better way than using tcpdump (via Net::Pcap) to achieve this?
Yes. Using libpcap (that's what you meant instead of tcpdump in that question) is a bad way to implement a TCP-based service, as you will have to reimplement much of TCP yourself (libpcap gives you raw network-layer packets), and the packets your program gets will also get delivered to the Internet protocol stack on your machine, so:
if there's nothing on your machine listening on the TCP port to which the other machines are trying to connect, the connection requests will get a RST from the TCP code and think the connection attempt failed;
if there is something on your machine listening on that port, it'll probably accept the connection, and it and your program will both try to communicate with the other machine, which will probably confuse its TCP stack and cause various bad and random things to happen.
It's not much better for UDP:
if there's nothing on your machine listening on the UDP port to which the other machines are trying to connect, the connection requests will probably get an ICMP Port Unreachable message from the UDP code, which may make it think the connection attempt failed;
if there is something on your machine listening on that port, it'll probably accept the connection, and it and your program will both try to communicate with the other machine, which will probably confuse its SIP stack and cause various bad and random things to happen.
IO:Socket will probably not give you raw packets, and that's a good thing; you won't have to implement your own IP and TCP/UDP stack. If your goal is to implement a redirect server on your machine, you have no need to receive raw packets; you want to receive SIP INVITEs with all the lower-level processing done for you by your machine's IP/TCP/UDP stack.
If you already have a SIP implementation on your machine, and you want to act as a "firewall" for it, so that, for some INVITEs, you send back a 302 redirect and prevent the SIP implementation on your machine from ever seeing the INVITEs in question, you will need to use the same mechanism that your particular OS uses to implement firewalls. There is no libpcap-like wrapper for those mechanisms, as far as I know.
I am still learning about SIP and all its protocols, specifically trying to integrate PJSIP into an iPhone application to make p2p calls.
I have a question about a peer 2 peer connection using PJSUA. I am able to
make calls perfectly to other clients on my local network by calling directly using the URI:
sip:192...*:5060
I am curious if this will work for
making direct calls to other SIP URIs that are not on the local
network without using server configuration - if not this way, is there another way of making p2p calls without server configuration?
thanks in advance,
You can make calls without server configuration, as a general principle, but something needs configuring. As mattjgalloway points out in the comments below your question, the most robust solution is a can of worms involving ICE which provides a kind of "umbrella" protocol for things like STUN.
Last time I touched this issue, I had the requirement that I couldn't use internet-based SIP servers to help. I came up with the idea of a registry of sorts: your client can define a bunch of "address spaces" with particular routing requirements. For SIP URIs in your LAN, you define no routing; for URIs in your company's VPN-accessed network, you define a route passing through your VPN connection; for everything else you define a route through your internet router.
By "define a route", I mean that when you place a call to a URI in some particular address space, you store what IP will go into a Contact header, what Route headers you might need, and so on.
Thus, the process of making a call becomes:
Look up in the set of address spaces for a match.
Ask that address space for the suitable bits needed to make a workable INVITE (appropriate Contact header details, Route headers, etc.)
Construct a normal INVITE, mutating as necessary for the previous step.
Send the INVITE as normal.
This essentially reproduces half of what ICE would give you, in a manually administrated form. "Half", because this ensures that one SIP agent can make calls such that the SIP routing all works. The missing half is you still need some kind of registrar somewhere, and each agent in your contact list needs to have the necessary setup to receive incoming calls. (If an agent's behind a NATting internet router, the router would need to either run a SIP proxy, or forward ports 5060, 5061 to a particular machine (which might be an agent, or a proxy serving the LAN's agents).
It is, indeed, a large can of worms.
The basic issue is to solve the problem of getting transport ports anywhere on the internet for multimedia traffic.
Many companies/experts have tried to solve this situation. A possible way out of is to buy a domain and setup a basic registrar using YATE or Asterisk on an address accessible from the internet and configure it to also use ICE as needed. Your iphone application at both ends could register automatically to it upon start. Then make P2P calls.
I am trying to make an application for iPhone that can listen for traffick on a specific network port.
A server on my network is sending out messages (different status messages for devices the server handles) on a specific port.
My problem is that when I make a thread and makePairWithSocket I block the port for others who want to send messages to the server, so I only want to listen to the traffic on a specifyed port and then check for specific heraders and then use those messages.
I know how to make the connection and talk to the server using write and read streams, but then I makePairWithSocket and block the port for all other devices on the network
Any one that has any suggestions on how to listen on a port in Objective-C without pairing with the server?
Thanks in advance
Daniel
Check out CocoaAsyncSocket. It gives you a nice and structured way (with delegates) to send and receive data... also with multiple clients. The documentation is quite good. project link
edit: Have a look at the AsyncUdpSocket class for a stateless UDP connection.
I think this requires network support well below the socket API level, perhaps at the hardware driver level, assuming the packets are even being routed to your device.