I am using csipsimple as sip client and asterisk server to set up call. Calls are made between 2 sip clients but voice is not getting transferred.
Calls are made between 2 sip clients using AMI.
I can give my asterisk cli log.
Can anybody please give me some idea to solve this issue?
Thanks
More info would be useful. First, make sure both clients are registered, and can use at least one common codec. In most cases, these aren't the problem. It's usually a NAT/Firewall issue. Are the two clients on the same subnet? Is there any firewall rules blocking the communication?
SIP signaling usually goes on udp:5060. But that seems working. Media is tricky. In each call, the ports for RTP audio changes, in the range specified in rtp.conf. This RTP traffic goes over UDP as well. By default it't 10000-20000.
If there is only routing done between the two endpoints, it should still be fine. NAT (Network Address Translation) is your main concern. Take a look at iptables, sip_nat_conntrack. To debug, use asterisk's sip set debug on command and look for the SIP headers and verify the correct IP addresses.
Related
I am puzzled...
My project is a PC connected to multiple micro-controller boards in an isolated network. So far the protocol has been UDP which is easy to deal with, has no particular client/server but has its obvious shortcomings of lost messages when things get busy.
The micro-controllers have fixed IP-addresses (set by dip switches), the PC SW has a list of them, sends at present UDP messages to each of them and they reply to the address they came from (i.e. the PC) with status and/or data.
My question is now that I switch to TCP instead of UDP, should the PC be the listening server with many clients (could be anything from 1 - 50), or should the micro controllers be listening servers the PC can connect to as client? Note: controllers have fixed/known addresses - the PC does not.
An additional concern is re-connection. The micro-controllers are external and may lose connection, reset or otherwise need to connect again.
Thanks....
should the PC be the listening server with many clients (could be anything from 1 - 50), or should the micro controllers be listening servers the PC can connect to as client?
That is a basic design question that we cannot answer for you. Likely, it's more practical for arbitrary devices to connect to a central server but that's not a given.
controllers have fixed/known addresses - the PC does not.
That might turn the previous question around.
The micro-controllers are external and may lose connection, reset or otherwise need to connect again.
That's something you need to put into your design - have TCP connections time out and reconnect. Usually, a finite-state machine is useful here. You should also consider whether you use a one-shot connect-transmit-disconnect similar to UDP (easier to implement) or a longer TCP session with multiple data transmissions (more efficient).
When call is made between two chan_mobile channels, all works fine.
But not when call is established between SIP and chan_mobile (through simple bridge).
SIP -> mobile is clear and fine with controllable packet size in sip.conf
Reverse direction voice going in too small packets causing near 180 kbits/s and loosing packets because of out of order on SIP phone side (if not enabled permissive mode on RTP).
So question is how to increase packets size originated by chan_mobile and going to SIP?
Setting like
allow=ulaw:100
not works in chan_mobile.conf
Chan mobile to chan_sip connection is via sln internal bridge ALWAYS.
You can control only outgoing packets(from your server sip to remote sip) by set
allow=g729:20
Or similar form. Please note, that codec should support length of packet you asking.
You have no any way control other side, so no way control inbound direction.
For more info see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
I know, how to write a C# application that works through a local network.
I mean I know, how to make my client-side application access my server-side application in a single local network.
But I wonder: How do such apps, as Skype, TeamViewer, and many other connect via global network?
I apologise, if this question is simple or obvious, but I couldn't find any information about this stuff.
Please, help me, I'll be very grateful. Any information is accepted - articles, plain info, books,and so on...
Question is very wide and I try to do short overview.
Following major difference between LAN (Local Area Network) and WAN (Wide Area Network):
Network quality:
LAN is more or less stable, WAN can be with network issues like:
Packet loss (you need use loss-tolerant transport like TCP or UDP with retransmits or packet loss concealment)
Packet jitter (interpacket intervals may differ a lot from sending part). Most common thing is packets bursts.
Packet reordering
Packet duplication
Network connectivity
WAN is less stable than LAN. So you need properly handle all things like:
Connection stale
Connection loss
Errors in the middle of the connection (if you use UDP for example)
Addresses:
In WAN you deal with different network equipment between client and server (or peers in case of peer-to-peer communication). You need to take in account:
NATs - most of the clients are behind NAT and you need to pass them through. According technics are called "NAT traversal"
Firewalls - may ISP has own rules what client can do or can't. So if you do something specific like custom transport protocol you may bump into ISP firewalls.
Routing - especially multicast and broadcast communication. In common case multicast is not possible to route. Broadcasts are never routed. So you need to avail this type of communication if you want to use WAN.
May be I forgot something. But these points are major. You can read many articles about any of them.
I'm trying to confirm a user a unplugged my embedded device from a network before performing some maintenance. I'm considering "ping"ing all IP address on my sub-net, but that sounds crude. Is there a broadcast/ARP method that might work better?
You can try a broadcast ping (this is from linux):
ping -b 255.255.255.255
Another option is to download Nmap and do a ping-scan.
You could use nmap. It's still crude, but at least it's using a tool designed to do it so you don't have to spend time on it.
If you can't get reliable link state information from your Ethernet device (which most chipsets should support these days, BTW...), sending an ARP request for each IP on your local subnet is a decent substitute. The overhead is minimal, and as soon as you get a single response, you can be sure you're still connected to a network.
The only possible problem I see here, is that if your device is on a /8 subnet, it can take quite a while to loop through all 4294967296 possible IPs. So, you may want to consider some optimization, such as only sending ARP requests for your default gateway, as well as all IPs currently in your ARP table.
If there's a peer you know you were connected to recently you could try pinging or arping that first. That could cut down on the traffic you're generating.
you could also run tcpdump -n to see what's active on the network too.
Not receiving any responses to ICMP pings or ARP requests is not a 100% guarantee that there's no network connection. For instances, there might be devices on the network that are firewalled off.
EDIT: May be you could access some lower-level information on your embedded device to check whether the network interface has its link up without actually sending any data.
Is there any chance that your device supports UPnP or Bonjour? Beside of the low-level protocols your should also have a look at these protocols which support some kind of plug-&-Play functionality. A UPnP device for example sends a message on the LAN before it is switched off (though, this doesn't help if it is just removed by unplugging it...).
From a Joel's post on Copilot:
Direct Connect! We’ve always done
everything we can to make sure that
Fog Creek Copilot can connect in any
networking situation, no matter what
firewalls or NATs are in place. To
make this happen, both parties make
outbound connections to our server,
which relays traffic on their behalf.
Well, in many cases, this isn’t
necessary. So version 2.0 does
something rather clever: it sets up
the initial connection through our
servers, so you get connected right
away with 100% reliability. But then
once you’re all connected, it quietly,
in the background, looks for a way to
make a direct connection. If it can’t,
no big deal: you just keep relaying
through our server. If you can make a
direct peer-to-peer connection, it
silently shifts your data onto the
direct connection. You won’t notice
anything except, probably, much faster
communication.
How do they change the server connection to a P2P connection?
It's pretty tricky and interesting. I'm sure I have some details wrong, but the overview is this:
The programs can already talk to each other through Joel's server, so they can exchange information with each other and Joel's server. Further, Joel has their external IP addresses, and they give joel information about their internal IP addresses.
They decide to try this hole punch technique. Computer A initiates a TCP connection with Computer B using B's external IP address. It won't go through, but what it does is tell's A's router that it needs to allow incoming packets from B on a given port.
Computer B does the same thing, but its message gets through to A since A's router opened a port/ip combination that matches what B sent (there's some port magic that happens here - this is non trivial, but doable).
B's router remembers that B initiated a connection with A on a given port and IP, and so A's packets now flow into B past their router correctly as well.
So it's actually pretty straight forward, but the implementation has details, especially regarding how ports are given to new TCP connections, and how NAT routers typically deal with TCP requests and how they map to external ports. These details are the interesting, and difficult, bit.
-Adam
I believe the simple version is that they drop the server connection and replace it with the P2P connection.
Something along the lines of:
Machine1 connects to copilot's servers.
Machine1 connects to copilot's servers.
Machine1 connects to copilot's servers.
Machine2 subsequently connects, and they begin screen sharing.
Machine2 opens a port intended for Machine1 to connect to.
Machine1 tries to connect to the now open port on Machine2.
If this connection is established:
The connection to copilot's servers is severed.
Data is instead transfered over the direct (P2P) connection between the two machines.
There is a technique called "Hole Punching" that works well with "Cone" NAT (Cone is a technical familly of router). That's not an 100% sure technique, today, it works well with UDP on about 80% of the router.
There is some implementations of library to realize Hole Punching: STUN (wikipedia)