Re: [Linphone-developers] Problem: ?eXosip: no ACK received during 20s: dropping call?

[IVAN FERNANDEZ]

Hi all,

Finally, after a deeper analysis of the issue, we’ve found a better solution for this problem, that looks a linphone issue after all.

The analysis has revealed that in the described scenary,  Linphone isn’t forming correctly the ‘SIP 200 OK’ message that notifies the call acceptance: the ‘contact’ field must include the ‘transport=tls’ sufix but the packet that Linphone is generating doesn’t includes this sufix, what is causing a port mismatch that causes not only this problem but any more.

In order to solve this issue we’ve had to modified the eXtl_tls.c file for appending this ‘transport=tls’ sufix to the contact field in the following way (within tls_tl_send_message function):

...

SSL *ssl = NULL;

   

   //*************************************************

   //the modification starts here:

char *msg;

char *contact;

char *newContact;

osip_contact_t *c;

char *result = NULL;

//we get the message

osip_message_to_str(sip, &message, &length);

//check if a 200OK message

msg = strtok(message, "\n");

if (strncmp(msg, "SIP/2.0 200 OK",14) == 0){

osip_contact_init(&c);

osip_message_get_contact(sip, 0, &c);

osip_contact_to_str(c, &contact);

//check if the ‘contact’ includes the ‘transport’ sufix

result = strstr(contact, "transport");

if (result == NULL){

contact[strlen(contact) - 1] = '\0';

newContact = (char *) osip_malloc(strlen(contact) + 16);

sprintf(newContact,"%s;transport=TLS>", contact);

/*OSIP_TRACE(osip_trace(__FILE__, __LINE__, OSIP_ERROR, NULL,

            "New Contact: %s\n", newContact));*/

//update the ‘contact’ field to include the missing ‘transport’ sufix

osip_list_remove(&sip->contacts, 0);

osip_message_set_contact(sip, newContact);

}

}

   //the modification ends here

   //******************************************************************

if (host == NULL) {

…

Any comments about this issue/solution?

Thanks and regards...

2012/8/23 Ramon Poca <address@hidden>

Then the problem is in the SIP routing setup, not on Linphone.
Breaking Linphone to fix whatever your Asterisk+opensips is doing
wrong seems an ugly fix.

If the dialog is not ACKed after a certain timeout, the UA should drop
the call. You should look at your opensips/asterisk logs to check if
it's being dropped.

> We use opensips as proxy to register the softphones and this opensips
> routes the call between the softphones throught an asterisk server.
> The problem with no ACK only appears if the called terminal is registered
> with TLS transport. Despite this lost ACK, the call is coursed without
> other problems...
> Regards
> 2012/8/22 Ramon Poca <address@hidden>
> > If no ACK is being received there might be some trouble in the call setup
> > (e.g. the SIP proxy is dropping it). What is the exact scenario?
> >
> > Ramon Poca - address@hidden
> >
> >


Ramon Poca - address@hidden

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