SIP stands for Session Initiation Protocol. It is provided by telepathy-rakia (or, as it was called: telepathy-sofiasip)
The second most common problem with making SIP calls is servers that throw away large messages (over 1500 bytes). To see if this problem is affecting you, run:
- killall telepathy-sofiasip ; SOFIASIP_PERSIST=1 TPSIP_DEBUG=all TPORT_LOG=1 /usr/lib/telepathy/telepathy-sofiasip
and look for INVITE or 200 OK messages being sent *repeatedly* that are larger than 1500 bytes (SIP clients will always ACK messages that get through, so if messages are being resent, it means they are being lost). You can shrink the size of packets by removing codecs (in /usr/share/empathy/codec-preferences, by setting id=-1) or waiting until we have implemented the ideas in https://bugs.freedesktop.org/show_bug.cgi?id=20135 and letting us do it for you.
The most annoying problem with SIP as a protocol is that SIP does not handle NATs very elegantly. Most SIP servers force you to send all media via their relays, so this problem is hidden from you as a user. If you aren't on one of these magical servers (e.g. you're in a corporate network that does clever things with VoIP phones) then you will have to play with the "discover STUN" and "STUN server" options until you are able to make calls. You will probably also find that different settings work depending on whether the client you are trying to call is behind the same NAT as you or not. Sorry about that. There's not a lot we can do without changing the protocol (there *are* plans to support ICE in SIP, which will solve all of these problems, but it may take some time).