Tom Vacek wrote: > James Cornell wrote: >> Tom Vacek wrote: >> >>> James Cornell wrote: >>> >>>> Tom Vacek wrote: >>>> >>>>> Have you tried Blastwave mplayer? I haven't tried it with many >>>>> DVDs, but it's been very good for what I have tested it on. >>>>> Note: Use "cdrecord -scanbus" to figure out the SCSI address of >>>>> your DVD drive, and then call mplayer like this: >>>>> /opt/csw/bin/mplayer dvd:// -dvd-device /dev/dsk/c3t0d0s2, but >>>>> replace the numbers 3, 0 ,0 in the last part with the address >>>>> given by cdrecord. The shock value when somebody sees you type >>>>> that is priceless. >>>>> >>>>> As for Skype, I believe there is a way to use the Linux version on >>>>> Solaris. You'll have to Google that, and it's not pretty. You >>>>> would do much better to use SIP, especially if you live in China, >>>>> and Ekiga (aka "Video Conference") works out of the box, even on >>>>> Linux nowdays. (For a proxy, Ekiga's own is good, and iptel.org >>>>> is great.) >>>>> >>>>> Tom >>>>> >>>>> >>>>> secret squirrel wrote: >>>>> >>>>>> seems like this is a good place to start moaning if you want >>>>>> skype for solaris >>>>>> >>>>>> http://forum.skype.com/index.php?showtopic=26856&st=20&gopid=1052331&#entry1052331 >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> desktop-discuss mailing list >>>>> desktop-discuss at opensolaris.org >>>>> >>>> Gizmo's proxy service also works. >>>> >>>> Download their client (Lin/Mac/Win) (There's a url somewhere) to >>>> register an account and install onto a computer with one of those os, >>>> virtualbox, or brand-z if you're feeling dangerous. >>>> Login to gizmo5.com for your sip number >>>> Set server to proxy01.sipphone.com >>>> Set username and auth username to your id number (Mine is >>>> 17472942049 if >>>> you wanna call me) >>>> Fill in your password >>>> Set stun to: >>>> stun01.sipphone.com (Ekiga.net should work equally despite them being >>>> different. >>>> >>>> Reference: >>>> http://support.gizmoproject.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=402&nav=0 >>>> >>>> http://forum.gizmo5.com/viewtopic.php?p=37399 >>>> >>>> Send e-mails at Gizmo's developers to make a Solaris client, they are >>>> probably more than willing, and are more open to the public than Skype >>>> who mirrors a traditional monolithic evil entity IMHO. >>>> >>>> I use ekiga too (sparcdr at ekiga.net) but it sometimes goes down >>>> giving me >>>> stupid errors like Authorization not allowed. (503) >>>> >>>> James >>>> >>> Ekiga's SIP proxy has some rough edges, but I've had excellent luck >>> with iptel.org. Also, SIP Communicator is a SIP UA that is written >>> in Java. The platform independent version works fine on OS, and it >>> has all the features of Skype: chat, voice, and video. With a >>> single exception,* I can't think of anything nice to say about >>> Skype. It hijacks your computer to proxy other people's calls >>> whenever it is running, and it probably does other things which we >>> don't know about. I tried their PSTN connection service back when >>> it was free, and I was totally unimpressed. As a PSTN gateway, the >>> quality was worse than an old dialup modem paired with >>> Asterisk/Zaptel. Their prices are lousy, too, and e911 is not >>> available even if you wanted it. All in all, one of the reasons I >>> like OS is that I can't install Skype, so if someone asks I get to >>> be the victim of evil proprietary software rather than a free >>> software curmudgeon. If there's ever a binary, I'm not sure where >>> I'll go. OpenBSD, perhaps, or Plan 9. That's probably safe. Sorry >>> about the rant. >>> >>> *Skype does use a wideband codec for non-PSTN calls, but few SIP UAs >>> have any wideband capability. >>> _______________________________________________ >>> desktop-discuss mailing list >>> desktop-discuss at opensolaris.org >>> >> I prefer GSM when I can use it, since I'm typically running things over >> a VPN with some latency overhead, while Speex unfortunately (Any of them >> for that matter) produce some less than steller results on some network >> setups. iLBC is also not very good, and neither is raw PCM. G.711 is >> decent though. I don't like how Skype hijacks either, though I do miss >> features in Skype which mirror what my private SILC has, encrypted chat >> that is. File sharing is also an issue, unless of course you have your >> own dedicated server (Or VServer) to transfer files. On many >> connections, the results of direct sharing using Pidgin with >> AIM/MSN/Yahoo isn't something you'd want to depend on. I'll try IPTel >> since you mentioned it. >> >> James >> > Does Skype do OK over a VPN? I believe they are using iLBC and its > wideband proprietary cousin iSAC. Perhaps a bigger issue is jitter > buffering, and I have yet to find a SIP softphone that does that > well. Often, the JB interpolations are as bad as the jitter. I've > never had a chance to do a serious test of Skype on a bad network. > SIP softphones are sitting ducks for feature bloat. From a networking > standpoint, chat and file transfer are simple extensions of SIP, but > it's a lot of work on the UI side. I would say that a lot of the > nuts-and-bolts aspects of a good softphone (ICE, jitter buffers, and > better codecs) have been overlooked in all the open source > implementations in favor of trying to match Skype's feature set. No Skype doesn't do great over a VPN unless you have less than 100ms latency to the remote system (Round-trip). For my friends in India, Sweden and AU, SIP works much better (Even with G.711).
Agreed with trying to match Skype's feature set. It's a lofty goal, when the main purpose is solely voice, which needs work in the quality department. For long latency, the generic open-source codecs fair well with the obviously less clear quality, but for those within the country, Skype is a much better case when running on unrestricted networks such as home DSL or cable lines and office T1s. There is serious jitter with Skype with 3g networks, which run an average (Lowest possible in most cases) of 160ms. SIP runs much better on these networks, with equal or less overhead (10kb/s up, 10kb/s down from my experience) or 80kbit raw with 8-16kbit overhead, about 176kbit total, within capability of being shoehorned into CDMA2000 or EDGE actually. Skype seems to run more around the 16kb/s range but is variable and depends on the link, and is probably affected by the peering system. I can't speak for Windows CE clients running on cellular networks, but I'm certain Skype doesn't fair very well. My friend who works as a contractor for Nokia (Network engineering in software) runs SIP on his S60-enabled smartphones (N93, N95 mainly) as well as Icecast (Using lame for mp3 encoding of a mpc controller) and seems to get good results. I think Skype performs better not only because of commercial codecs in local countries, the peering system reduces long distance latency overhead. James
