The only point you make that reflects reality is 2. But I'll address them all.
1. SIP has a wide range of codec options with varying quality (and bandwidth usage). Many open codecs have much better quality than Skype, many have worse. I get comments to that effect ("Wow! Your voice is so clear!") from Skype users when I call them landlines via Ekiga and a SIP callout service (diamondcard.us). Your experiences with poor voice quality have nothing to do with Skype vs SIP and everything to do with the codecs supported by both endpoints. Of course, Skype has its own proprietary codec, and it is decent, but is not the best by any means (for either bandwidth or quality).
2. The big problem with SIP is NAT. SIP *would* be extremely easy to set up if it didn't use multiple ports and also have to deal with NAT. (SIP clients are already cross platform and easy to install.) I wince as hundreds of first time users on the Ekiga forum are turned away because of frustrating NAT issues. Solutions to this are:
a) VPN. I use SIP with family and friends via a private VPN that I set up (openvpn). There are fine VPN programs for Windows as well (including openvpn). This bypasses NAT issues for those on the VPN.
b) IPv6. No NAT issues when both ends have an IPv6 address.
c) IAX - Inter Asterisk eXchange protocol - this multiplexes the many ports used by SIP and RTP/sRTP into a single UDP port that traverses NAT with ease. Only a few clients support this at present (Ekiga does not).
d) sip proxy - the technically savvy can run a sip proxy on their gateway linux server. This essentially provides smart protocol aware NAT for multiple simultaneous SIP calls through a single gateway IP. The less technically savvy can purchase a router like box (personal PBX) for $500 - $1000 that provide sip proxy for internal smartphones, and some telco ports for trunks and POTS extensions. (The box contains embedded asterix or equivalent.)
3. Using one of the many free SIP conferencing services is easy as dialing the same room #.
4. SIP does *not* support screen sharing, not should it - there are so many good ways to do screen sharing. Bundling it with Skype just rubs in the security risk that Skype presents - you are giving all other Skype programs on machines all over the internet the equivalent of ssh access to your machine. Anyone that "cracks" the Skype binary has free access to all those Skype using machines. (In this case, I sincerely hope that Skype is really really difficult to crack...)
I read a bit about Jingle. It still uses RTP for the actual voice/video, so it has the exact same NAT problems as SIP (that STUN does not solve except in very simple cases). The only open protocol (other than IPv6) I am aware of that completely addresses NAT is IAX - where SIP and RTP are encapsulated in a single UDP port.
Authored Comments
The only point you make that reflects reality is 2. But I'll address them all.
1. SIP has a wide range of codec options with varying quality (and bandwidth usage). Many open codecs have much better quality than Skype, many have worse. I get comments to that effect ("Wow! Your voice is so clear!") from Skype users when I call them landlines via Ekiga and a SIP callout service (diamondcard.us). Your experiences with poor voice quality have nothing to do with Skype vs SIP and everything to do with the codecs supported by both endpoints. Of course, Skype has its own proprietary codec, and it is decent, but is not the best by any means (for either bandwidth or quality).
2. The big problem with SIP is NAT. SIP *would* be extremely easy to set up if it didn't use multiple ports and also have to deal with NAT. (SIP clients are already cross platform and easy to install.) I wince as hundreds of first time users on the Ekiga forum are turned away because of frustrating NAT issues. Solutions to this are:
a) VPN. I use SIP with family and friends via a private VPN that I set up (openvpn). There are fine VPN programs for Windows as well (including openvpn). This bypasses NAT issues for those on the VPN.
b) IPv6. No NAT issues when both ends have an IPv6 address.
c) IAX - Inter Asterisk eXchange protocol - this multiplexes the many ports used by SIP and RTP/sRTP into a single UDP port that traverses NAT with ease. Only a few clients support this at present (Ekiga does not).
d) sip proxy - the technically savvy can run a sip proxy on their gateway linux server. This essentially provides smart protocol aware NAT for multiple simultaneous SIP calls through a single gateway IP. The less technically savvy can purchase a router like box (personal PBX) for $500 - $1000 that provide sip proxy for internal smartphones, and some telco ports for trunks and POTS extensions. (The box contains embedded asterix or equivalent.)
3. Using one of the many free SIP conferencing services is easy as dialing the same room #.
4. SIP does *not* support screen sharing, not should it - there are so many good ways to do screen sharing. Bundling it with Skype just rubs in the security risk that Skype presents - you are giving all other Skype programs on machines all over the internet the equivalent of ssh access to your machine. Anyone that "cracks" the Skype binary has free access to all those Skype using machines. (In this case, I sincerely hope that Skype is really really difficult to crack...)
5. SIP is also a really good chat system
6. Video works great too.
I read a bit about Jingle. It still uses RTP for the actual voice/video, so it has the exact same NAT problems as SIP (that STUN does not solve except in very simple cases). The only open protocol (other than IPv6) I am aware of that completely addresses NAT is IAX - where SIP and RTP are encapsulated in a single UDP port.