I'd like to get in touch with Fanjita team /Noobz team for some improvements and completion on Furikup Video and Voice psp-phone.
My issues:
- I'd like to develop a full 3.90 cf integrated SIP Phone adapting Furikup source code by Fanjita team, of course sharing the results and sources
- I've already realized a successful Video and Voice SIP phone between two PSPs
- I've developed a new Furikup without 1.50 user mode prx.
Here are issues to avoid 1.50 user mode prx:
- use
Code: Select all
sceUtilityLoadNetModule(1);
sceUtilityLoadNetModule(3);
connect_to_apctl(MyVarIPConnection)
- change all threads calls
- in codec.c of SIPEngine do not use
Code: Select all
stunParseServerName(config_get_stun_server(), &dest);
because in stun.c of ortp library
Code: Select all
gethostbyname
and use instead the resolver:
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void stun_server_resolver(char *value)
{
int rid = -1;
char buf[1024];
struct in_addr addr;
char name[1024];
while(1)
{
if(sceNetResolverCreate(&rid, buf, sizeof(buf)) < 0)
{
printf("Error creating resolver\n");
break;
}
printf("Created resolver %08x\n", rid);
if(sceNetResolverStartNtoA(rid, value, &addr, 2, 3) < 0)
{
printf("Error resolving %s\n", value);
break;
}
printf("Resolved %s to %s\n", value, inet_ntoa(addr));
break;
}
if(rid >= 0)
{
sceNetResolverDelete(rid);
}
}
I'd like to share thes issues with Fanjita team or others in order to finalize a Video and Voice PSP-SIP Phone with full functions and good quality.
Thanks for feed back and contacts.
My best to all