I have developed a small voip client application in C using sofia library and based on the sofia-sip client example.
Evertyhing works fine but one thing, if my Ip address changes once the application is running sofia doesn't update itself. I'll explain myself:
When I execute my voip application (running in Linux) sofia takes the current Ip address and the application will accept all the incoming calls to that address.
If the IP of the device where the application is running is changed (updating network config of Linux) i'm not able to update sofia in order to work with the new ip, therefore any incoming call is not received.
I've tried with ssc_set_public_address, which updates ssc_address and set the new value in nua using nua_set_params:
/**
* Sets the public address used for invites, messages,
* registrations, etc method.
*/
void ssc_set_public_address(ssc_t *ssc, const char *address)
{
if (address) {
su_free(ssc->ssc_home, ssc->ssc_address);
ssc->ssc_address = su_strdup(ssc->ssc_home, address);
nua_set_params(ssc->ssc_nua,
SIPTAG_FROM_STR(ssc->ssc_address),
TAG_NULL());
}
}
After doing it, it looks like the internal ip address has changed (if I read it using get_params I get the new one), however something is missed because no incoming call will be received.
In order to work with the new IP I need to reboot the whole application.
Does anybody know how to really update the sofia IP without rebooting the application?
Thank you.
Related
I want to connect two F746ZG boards so that they can communicate via TCP. I am using the STM implementation of LwIP with the netconn API. The IP address is supplied via DHCP, but it is always the same address. Also, the address matches the expected value. The problem I am facing is that the client seemingly can't establish a connection. I am binding the connection to port 8880. Since I ran into this issue, I have written a debug client that should just periodically send a predefined message to a server. Here is the code for the client:
static void tcpecho_client_thread(void const *arg)
{
struct netconn *xNetConn = NULL;
err_t bind_err, connect_err;
char* b_data = "OK"; // Data to be sent
uint16_t b_len = sizeof ( b_data );
IP4_ADDR(&local_ip, IP_ADDR0_CLIENT, IP_ADDR1_CLIENT, IP_ADDR2_CLIENT, IP_ADDR3_CLIENT);
IP4_ADDR(&pc_ip, IP_ADDR0_PC, IP_ADDR0_PC, IP_ADDR2_PC, IP_ADDR3_PC);
xNetConn = netconn_new ( NETCONN_TCP );
if (xNetConn != NULL){
bind_err = netconn_bind ( xNetConn, &local_ip, TCP_PORT_NETCONN );
if(bind_err == ERR_OK){
// Try to connect to server
for(;;){
connect_err = netconn_connect ( xNetConn, &pc_ip, TCP_PORT_NETCONN);
if (connect_err == ERR_OK){
// We are connected
while(1){
BSP_LED_On(LED1);
netconn_write(xNetConn, b_data, b_len, NETCONN_COPY);
vTaskDelay(1000); // To see the result easily in Comm Operator
}
}
}
}else{
// Failed to bind the connection
BSP_LED_On(LED3);
}
}else{
// Failed to allocate a new connection
BSP_LED_On(LED3);
}
}
When I debug this, netconn_connect never manages to actually connect to something. Since I am able to ping the board and get a response, I am confused, what is going wrong here. I have tried to use Hercules to set up a TCP server on my PC so that the board can connect to that, but that also doesn't work. Using Wireshark, I can see the responses to my ping command coming in, but I don't see anything that would indicate the board trying to connect to my PC.
I have tested the corresponding server on the second board, but that runs fine. I can connect to it with Hercules and send data, so I doubt there is anything fundamentally wrong with the LwIP stack.
What I could guess is that I messed up the netconn_bind, I am not 100% sure what IP you are supposed to bind the connection to. The way it currently is, is how I read the documentation. For the server, I have bound it to IP_ADDR_ANY. Besides that, my implementation mostly matches with the examples you can find online (e.g. LwIP Wiki).
I have figured out the problem. After I delete the netconn_bind call, everything works fine for me.
I have the following service.
Spring boot 2.5.13
Camel 3.18.0
JMS
I want to use an embedded ActiveMQ Artemis, standalone ActiveMQ Artemis, and IBM MQ.
I've managed to get all 3 running and connecting, but one thing I cant figure out is the JMSReplyTo option.
Running locally with embedded broker:
This runs fine. I can write a message to the queue and a response is send to the JMSReplyTo:
public void sendRequest(){
ActiveMQQueue activeMQQueue = new ActiveMQQueue("RESPONSE_QUEUE");
jmsTemplate.convertAndSend("REQUEST_QUEUE", "Hello", pp -> {
pp.setJMSReplyTo(activeMQQueue);
return pp;
});
}
Via ActiveMQ Artemis console:
This is where the inconstancy comes as the Object received is an ActiveMQDestination which makes setting the CamelJmsDestination much more involved.
Am I wasting my time here? Should I just grab the queue name and construct the uri manually? Or I am missing some logic as to how this works? Or maybe I'm not using the Artemis console in the correct way?
.setExchangePattern(ExchangePattern.InOut)
.setHeader("CamelJmsDestination", header("JMSReplyTo"))
When using javax.jms.Message#setJMSReplyTo(Destination) you have to pass a javax.jms.Destination which must implement one of the following:
javax.jms.Queue
javax.jms.TemporaryQueue
javax.jms.Topic
javax.jms.TemporaryTopic
In order to reproduce this semantic via text in the web console of ActiveMQ Artemis you need to prefix your destination's name with one of the following respectively:
queue://
temp-queue://
topic://
temp-topic://
So when you set the JMSReplyTo header try using queue://RESPONSE_QUEUE.
When your application then receives this message and invokes getJMSReplyTo() it will receive a javax.jms.Queue implementation (i.e. ActiveMQQueue) and then you can use getQueueName() to get the String name of the queue if necessary.
I am using Pjsip library to register a Sip account to Sip server
/* Register to SIP server by creating SIP account. */
{
pjsua_acc_config cfg;
pjsua_acc_config_default(&cfg);
cfg.id = pj_str("sip:" SIP_USER "#" SIP_DOMAIN);
cfg.reg_uri = pj_str("sip:" SIP_DOMAIN);
cfg.cred_count = 1;
cfg.cred_info[0].realm = pj_str(SIP_DOMAIN);
cfg.cred_info[0].scheme = pj_str("digest");
cfg.cred_info[0].username = pj_str(SIP_USER);
cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
cfg.cred_info[0].data = pj_str(SIP_PASSWD);
status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);
if (status != PJ_SUCCESS) error_exit("Error adding account", status);
}
This working fine when my board only has one network interface, but when i run it on a new board which has many network interface, it always register to Server via interface have smalless IP address value. I read in this link: https://trac.pjsip.org/repos/wiki/FAQ#multihomed And realize that Pjsip always chooses a network interface to register by its algorithm. And i tried to bind my application to a newtork interface too like this: https://unix.stackexchange.com/questions/210982/bind-unix-program-to-specific-network-interface but it not work too. I need to choose network interface manually in the code, how can i do it ?
Search for parameters named bound_addr in pjsip configuration - there should be (unless something changed with updates) one in transport configuration and one in rtp configuration. This is binding by interface address, not interface name, but it might be a starting point for modifications if necessary.
I have an issue of an if statement not passing whilst my system gatt connection is not made.
Context
I have a BLE system using a NRF52840-dk board programmed in C. I also have a mobile application which, communicates with this board via a Gatt connection. I have a single service with a single characteristic. I write to this characteristic from my mobile application and, from this do some processing. At the moment I can send over a timestamp and begin storing data. However, I need to then send data back to my mobile device by this connection.
So what I have is a command to be sent from the phone to ask for some data. This should then send data back to the phone by changing the characteristic value.
Before I can change the value I need to see if the command has been issued. However, due to the priorities and constraints of the device I need to do this processing in the main function not in the BLE interrupt that I have done my time stamping in. This is due to the data I will be transmitting eventually will be large.
My issue however is, I receive the command to send some data back to the phone and update a global int value (changed from 0 to 1). Then in my main loop test this value and, if it is 1 write to the terminal and change the value back. I would then use this point of the code to run a function to send the data.
But this statement does not pass.
This is my main loop code
if(GATT_CONNECTED == false)//This works!
{
//Do some functions here
}
else if (GATT_CONNECTED == true)// GATT_CONNECTED = true
{
NRF_LOG_INFO("Test1 passed");//Testing variable this does not print
if(main_test == 1)
{
NRF_LOG_INFO("Test2 passed");//This does not print either irrelevant of value
main_test = 0;//False
}
idle_state_handle();
}
I don't know if the issue is the way I have defined my variable or due to interrupt priorities or something like that. But, when my Gatt connection is made the loop of (GATT_CONNECTED == true) does not seem to process.
My variable is defined in another file where my GATT connection is handled. The GATT connected variable is handled in main. my main_test variable is defined in another c file as int main_test = 0;. In the header declared as extern int main_test;.
I know the GATT_CONNECTED variable works as I have code in it that only runs when my gatt is not connected. I have omitted it for simplicity.
Any ideas,
Thanks
Ps Hope you are all keeping well and, safe
Edit
Added code for simplicity
main.c
bool GATT_CONNECTED = false;
int main(void)
{
Init_Routine();
while(1)
{
Process_data();//This runs if the gatt is not connected if statement inside
if(GATT_CONNECTED == true)//This does not run true when the gatt is connected
{
NRF_LOG_INFO("check gatt connectedpassed");//Testing variable.
nrf_gpio_pin_set(LED_4);//Turn an LED on once led 4 does not work
}
idle_state_handle();
}
}
I have several embedded machines listening and streaming rtp audio data to a multicast group. They are connected to a smart managed switch (Netgear GS108Ev2) which does basic igmp snooping and multicast filtering on its ports, so that the rest of my (W)LAN doesn't get flooded.
At start everything works fine for about 500-520 seconds. After that, they don't receive any more data until they leave and join the group again. I guess the switch is "forgetting" about the join after a timeout.
Is there any way to refresh the group membership, i.e. letting the switch know, that there ist still someone listening, without losing packets?
System info:
Arch: blackfin
# cat /proc/version
Linux version 2.6.28.10-ADI-2009R1-uCBF54x-EMM
(gcc version 4.3.3 (ADI) ) #158 PREEMPT Tue Jun 5 20:05:42 CEST 2012
This is the way multicast / the IGMP protocol works. A client has to join the group periodically by sending a Membership Report or it will be assumed that he has left the group after some short timeout. However, those reports are usually sent only when receiving a Membership Query from the local multicast router. Either your clients don't receive the query or don't respond with a report.
Try to use a tool like wireshark in order to see which IGMP packets are sent through your network.
You need an IGMP querier to send the Membership Queries, as was already explained by scai.
If you can't configure your router to do that, you can use one of your computers. Seeing how running a full multicast routing daemon would be overkill (and I've never done that), I suggest you try to abuse igmpproxy.
First create a dummy upstream interface (this is not persistent!):
ip tap add dev tap6 mode tap
Write igmpproxy.conf:
# Dummy upstream interface.
phyint tap6 upstream ratelimit 0 threshold 1
# Local interface.
phyint eth0 downstream ratelimit 0 threshold 1
# Explicitly disable any other interfaces (yes, it sucks).
phyint NAME disabled
...
Finally start igmpproxy (as root):
igmpproxy -v /path/to/igmpproxy.conf
If your embedded devices are running linux, you need to turn off the reverse packet filter on them or they won't respond to group membership queries. In that case the upstream switch will assume there is no-one listening to that multicast and switch it off.
I had same problem, multicast on wifi was lost after 260 seconds, I solved it with my application by adding AddSourceMembership on socket.
private void StartListner(IPAddress sourceIp, IPAddress multicastGroupIp, IPAddress localIp, int port)
{
try
{
Socket socket = new Socket(AddressFamily.InterNetwork, SocketType.Dgram, ProtocolType.Udp);
IPEndPoint localEndpoint = new IPEndPoint(localIp, port);
socket.Bind(localEndpoint);
byte[] membershipAddresses = new byte[12]; // 3 IPs * 4 bytes (IPv4)
Buffer.BlockCopy(multicastGroupIp.GetAddressBytes(), 0, membershipAddresses, 0, 4);
Buffer.BlockCopy(sourceIp.GetAddressBytes(), 0, membershipAddresses, 4, 4);
Buffer.BlockCopy(localIp.GetAddressBytes(), 0, membershipAddresses, 8, 4);
socket.SetSocketOption(SocketOptionLevel.IP, SocketOptionName.AddSourceMembership, membershipAddresses);
try
{
byte[] b = new byte[1024 * 2];
int length = socket.Receive(b);
}
catch { }
}
catch (Exception ex)
{
logger.Error("Exception: " + ex);
}
}