Still no SIP?
 

I looked around a bit in the usual places but it seems there's still no simple SIP-compliant VoIP-application out there. Or did I miss something?
From what I understand, Gizmo locks you in with one provider (thats how I read it) and offers functionality I wouldnt need. All I want is to use one of my 2 existing SIP-accounts with my 770. Can I?

no sipping
 

I feel your pain. I've been working with asterisk end solutions and would give anything for a SIP client on the 770. If you run the regular version of Gizmo you can use a second provider and there are details for using it with a SIP provider. But there seems to be no info for the 770

After much googling I found a few mentions of SIP on the 770, such as Tapioca, etc. But it doesn't appear to be anyone actively working on it. Go figure.

real SIP solution requested!
 

You're right. No real SIP solution available for our prefered device :(
The 2nd account facility is available for version 2 of Gizmo Project: the last release for N770 is still version 1 despite the new release posted last week... what a pity!

Thank you - now at least I know that a) I'm not the only one who looks for SIP and b) I really did not miss an application.

Gizmo, btw, would not be what I'm after even if it had this "wnd account"-thing. I always feel its too "big" (I don't want to use the word "bloated" here because Gizmo seems to offer good features for its size, but they're no features I need).

I would feel very comfortable with something like Linphone. Linphone can even be run as a command line application and could be further stripped down by not including the part that does the video conferencing.

Can somebody try to port it? Christmas wuld be a nice release date :)

Check this out. I haven't tested it myself but it is a command line SIP client for 770 implemented using Nokia's own Sofia SIP stack.

- Teemu

Thanks for the link. I've followed the instructions about the repository but I can't find any Communication category nor sofsip app after having updated my list of packages... any clue? :confused:

Put a / at the end of the repository url : Web Address: http://jonek.hexbox.de/maemo/

Also, the command has an underscore instead of a dash: sofsip_cli --media-impl=fsgst

Thank you for this, but even with this extra slash, it still soesn't work: no "Communication" section, no sofsip entry in "All" section... Any new clue? TIA.

Quote From Jonek's page :

Web Address: http://jonek.hexbox.de/maemo
Distribution: mistral
Components: user

It worked for me.
Do you have other Application Manager issues?
Check the log.

Nothing special in the log...
Well, I've downloaded the package directly from the web repository, so it's OK now except that I still don't know what's wrong w/ my Apps Manager :(
Anyway, for what I've read and tested, sofsip doesn't work w/ my ISP (Free, in France, freephonie.net as domain). So I'm still looking for a SIP app for my N770 :rolleyes:

There is an error when i want to call someone.
i launch sofsip-cli with fsgst implementation and :
sofsip> UA: INVITE: 488 No vocodeur intersection

I tried a few things with Jonek's help but to no avail !
It looks like sofsip cannot negociate a codec with the french sip provider "free" !
I'll try a few other things in the meantime but
I'm waiting for tapioca with sip (telepathy)

Fred

I keep checking on Tapioca but there never seems to be any updates on the SIP portion of it. So I'm not sure about holding out any hope for that either. I'm just pretty surprised that if they were able to cram google talk in there, that they couldn't have put some other SIP solution in there as well. After all, if you google for 770 and SIP you get all kinda of hits with google talk in there. Lot of help that is.....

I've seen some that reference a third party you can register with, and use google talk and some sort of weird mis-mash. But thats not really what I want either. Just a simple, SIP client would be just about what I want. *sigh*

Kev

Ignore this posting, see below - it works.

Thx, I got it working - kind of. I can answer incoming calls, but I'm unable to make an outgoing call.
There are a lot of GLIB debug messages which I cant interpret; the one that repeat are "No RTP destination available" and "Unresolvalble destination".

I'm using the SIP-URL the I type into all of my softphones, so I think there's nothing wrong with it. Pity :(

I was able to get both incoming and outgoing working, but that text interface is pretty harsh. What about dtmf tones? It makes a barely usable client but I suppose thats something. What I'm really wondering if the CLI is there, and the libraries as well, couldn't something like this be done in ruby, or even a curses interface? Something needs to keep it registering and on the server like a real client. If I had any real code skills at all I'd be doing something. But a bad package manager still manages to stump me for days. So it appears I'm relegated to waiting on someone to release a better SIP interface.......

I have to correct my own message so it doesnt scare off others :)
It does work on my 770. The Problem was not with the Software but with my router being misconfigured (I had done all the ugly port forwarding stuff months before in a situation when it was safe to assume that only one PC in the network would ever run a SIP client).

A Telepathy (IM/VOIP framework user on the 770, see http://telepathy.freedesktop.org/) backend for SIP is being worked on by the Nokia Research team who works on Sofia SIP. See my post to the maemo-devel list for more details: http://thread.gmane.org/gmane.comp.h...784/focus=5794

Re: Still no SIP?
 

Hi!

I see that several people have managed to get sofsip-cli to work with nokia 770. I have just purchased a nokia 770 and I am trying to get sofsip to work over ipv6. sofsip is ipv6 enabled and I am able to make a call to another PC-Based ipv6 sip client (Kphone).

When i call kphone, it rings and i can answer the call. However, I cant hear any sound. Just want to inquire if you guys did any other configuration appart from what is mentioned on http://jonek.hexbox.de/?p=43.

I really want to make this sip over ipv6 work. Any assistance would be appreciated.

shariq

Re: Still no SIP?
 

hello, im using this plugin to the bult in messenger client on N800 so it Support
SIP, and it rely works :) http://rtcomm.garage.maemo.org/

Re: Still no SIP?
 

You can also wait for IT2008 instead of using the RTCOMM beta, if you'd like.

Re: Still no SIP?
 

Is there no hope for the 770 under 2006?

Re: Still no SIP?
 

But the Sofia SIP used in RTCOMM is still buggy. It won't work over a VPN connection for example.
And from what I've read in the bugzilla on it it's not going tobe fixed any time soon, not even in IT2008.

Re: Still no SIP?
 

Actually maximCT on #maemo said it worked over PPPoverSSH and over ssh -w (ssh tunnels)... But i don't have any more info for now.

I'm having the rtcomm client use the wlan0 ip instead of the tun0 IP! I'll see if i can fix this with the ip(8) command...

Re: Still no SIP?
 

+1 from me

Gizmo is a fine solution meanwhile now they allow alternative SIP-accounts to be used, but it doesnt integrate with the contacts and its - well, simply too much. If SIP could be integrated in the IM-framework on the 770, too, I'd love it.

Re: Still no SIP?
 

When people talk about sip they usually talk about pc to pc, or to other lan devices. In Norway we got televoip that give voip over sip but where you call to other phones. The settinhs that televoip got is same as freeworldialup. So are there any client like gizmo that got same settings as FWD. did not find any settings like that in gizmo.

Re: Still no SIP?
 

IBICO, which settings exactly are you talking about?

I use 2 sip-accounts with gizmo that allow me to call any landline/mobile phone, and i didnt have to tell gizmo more than account/server/password, IIRC.

Re: Still no SIP?
 

Auth name, passwrd, server (sip.televoip.no), server port (5060), and maybe stun server with port. not sure about codec.