Puppy Phone - VOIP using SIP

[smokey01]

Ah ok.. so we need to configure our routers for the correct ports eh? What port numbers?
I guess this is why I have success with 3G but none at home through the router.

Attached is a tool that will help you figure that out.

[DPUP5520]

Ah ok.. so we need to configure our routers for the correct ports eh? What port numbers?
I guess this is why I have success with 3G but none at home through the router.

One other odd thing I found out too was that while connected through my router and forwarding the proper ports I still had to connect to a call and disconnect a few times to get good quality for some reason, however when connected using an encrypted connection using my proxys in puppy crypt I did not have this problem as every call went through properly I'm still trying to figure out why.

[nooby]

Guys I would love to have this too but the problem for me
apart from the tech side of things to set levels and ports and
routers and such

this is the problem. I never shut up I just talk to much

[DPUP5520]

nooby the big problem on my end I think was that I am using a router that's just about brand new so there were a lot of configuration settings on it and it was set up to be pretty secure from what I was looking through, I had never really paid attention to the security settings since I run a pretty hardened OS as it is so I never worried about the router itself.
It seems most people didn't/don't have the same problems I've been having and the main setup for Psip is pretty straight forward with the registering and setting up the account and stuff. As I said all my stuff is pretty brand new (I like to have all the new toys to play with) so it seems that was most of the problem.
The biggest thing was the setting on my router sip ALG was disabled which needs to be enabled for Psip to work properly through a router if it has that option.

[nooby]

I have three Dlink routers and them are named D-link 524, 624 and 615
I am sure of them is set up like yours then to compete on the market.

Them usually try to live up to the standard on the market to be able to sell.

So I thank you for the tip on to look at such if the audio act up.

Smokey told that you guys talk without you logged in.
I guess it is about that he see your IP while if you are
logged in then he see your iptel name or your ekiga name?

I mean I should join both of them then and use same
username on both? That way I can join in with anybody?

But my problem is that I talk too much and it will only end in
embarrassment for everybody. But I do long for to chat though.

[DPUP5520]

Yes we did talk for a while when I was logged out and still trying to figure out my issues, you can only see your internal IP address when not logged in in other words he saw 192.1**.1.10* which is one of my internal IP addresses. You don't really need two for here from what I've seen it seems on one account you can talk to anybody, for example on my ekiga account I can call into the iptel conference room or talk to smokey01 who is on iptel. I set up both an iptel and an ekiga account simply because I had originally chosen ekiga and thought that might be the source of my problems so I made another one with iptel which im using as my main one now. And yes if you are logged in they will only see your name and not your IP address.
For some odd reason while I was using the ekiga softphone and did not log in I tried to call smokey01 and he could see an external IP address however it was not mine but in my area which was a bit strange.

[nooby]

Thanks Dpup
Is it like Gmail and their chat when I allow mmyself to be visible
and all those who are in my contact list if them allow themselves
to be visible then as soon as I switch on the power in the morning
and boot up into puppy and log on into gmail then my "contacts"
know that I am online and them can make a chat call if them so like.

[DPUP5520]

Yes

[DPUP5520]

Actually you bring up a good point nooby. Is there anyway to hide yourself once you are logged in or is logging out the only way?

[nooby]

Hope I am not derailing too much now but the owner of Facebook
says that the days of Anonymity is over. Deal with it.

I hope him is wrong because if a wife is beaten by the Hubby
ans she has to find a way to get safe from him how on earth
can she do that if Facebook gives her away?

Some kind of anonymity is also required for those living in
oppressing regimes.

And if one work in a company that behave bad and one want
to tell the media about it without getting fired and branded as
a whistleblower for the rest on ones life.

okay back on topic.

I guess as soon as somebody has the range of IP for you
then them can always guess you are one of the 500 that
is up that morning. Here in Sweden the Authorities has told
the IPS to give us same IP each morning so them can follow
us around more easy so one need to force a change and then
them try to change it back. I've given up on it.

Them want us to be known. I trust it has to do with the Pirates
of Movies and Music and Books them want easy way to keep track.
So I decided to stay outside of Torrent and such.

There I go again derailing. So you guys talk over IP phone

[Lobster]

For those not in the UK conference call is at the usual time
In the UK the clocks go back - so we are now on GMT (without the summertime)
Next conference call on Sunday October 30 2011 at 1200h GMT

[smokey01]

Time zones explained.

http://www.bbc.co.uk/news/world-12849630

[Lobster]

Sorry I did not make the conference call this week guys.
How did it go?
Next one Sunday 6th November 12:00 AM GMT

[DPUP5520]

I tried to make it in however was unfortunately incapacitated at the time.

[nooby]

My problem is that puppy phone just locks when the setting shall be closed
I just locks do nothing.
I will try the one in midnight sun to see if same happens
Very important info from Dpup

The biggest thing was the setting on my router sip ALG was disabled which needs to be enabled for Psip to work properly through a router if it has that option.

I trust my Dlink 524 is set up like that too and I have forgotten how I log into that one. Too long time ago for my poor memory to remember and
I most likely wrote it down on a ad paper and throw it away later.
Not realizing anything important on it.

Old psip already installed worked good. The new shiny one I'm too stupid
on hopw to set up.

old one asked nothing about firewall it just worked while the new one
asked about firewall and I had no idea what one should answer
and when I tried to go online it just tried and tried and never managed to do it.

I will be away now for some 3 hours or so.

Thanks for all the cheer up. Nice talking to you guys.

[Stripe]

Hi all

I have been reading up on possible (or unprobable) reasons for Drop Out Syndrome (which seems to be happening more frequently at the moment)

One possible cause could be that someone is interfering by unfair means (which I will not name so that no one can get any ideas)

I know that rtp traffic can be encrypted from the options but it has to be sent over UDP and anybody could still see the packets being transferred to and from the sip clients (even though they can't listen to the conversation they could still cause problems.)

could we try to use TCP for sip and encrypt it using TLS to prevent the traffic being seen and still have the RTP traffic encrypted to prevent eavesdropping. (and rule out another possible cause of the Drop Out Syndrome.)

A vpn could be an alternitive, but that would require a static ip or using a free domain (like dyndns) so that can be discounted

hope the above makes sense (no doubt dpup5520 will correct me if needed )

Cheers

Don

[gcmartin]

..

Good points to consider.

The following ONLY shares info on your VPN topic you raise.
On the VPN, I'm not sure how this would work because VPN is an end to end technology originally design to provide secure connections between two "pre-agreed" end-points.

Thus, the VPN architecture. It is most often used for "LAN to LAN via I-net" (like from a branch office to a main office, securely) and from "individuals to LAN via I-net" (like for a sales person connecting in to the home office, securely). There is the provision for a individual to individual as well.

A VPN is an individual connection. imagine "you and they" constructing a "tunnel" between you and the specific person you want to connect to. And, the tunnel MUST be maintained on each end. If one loses his tunnel connection script, they won't be able to use the tunnel. And, if you crafted a "generalized" tunnel in which everyone knew how to connect to you, you would defeat your objective for privatization should the tunnel information got out....right?

Your concern for security would most likely be served if you and the person whom you want to have a secure tunnel with work together on your personal tunnel(s). Then whatever you do in the tunnel would be secure....not just voice, but everything. Further, you could set up your site to be a VPN server to provide multiple connections to you by sharing the technology to do so with each user, if that meets your needs.

Hope this helps.

[gcmartin]

Just thought about my prior comments.

Question
Is Puppy Phone aimed at providing Voice calling services between PC users via SIP clients or SIP devices as a replacement to using my "Plain Old Telephone system" telephone to call someone OR is the mission something a little different?

I want to make sure I understand the mission. Thanks in advance.

[Stripe]

Hi all

have just discovered that if you have "secure voice call" checked
(encrypted by srtp) it will log in with iptel but will not let you call any of the "utilities" music, conference, etc.

gcmartin
As far as I am aware its just for pc to pc calls replacing the telephone and we are trying to help iron a few bugs out, (psst dont tell anyone but I think it is secretly being used by lobster as a black ops tool for world domination)

and remember "loose tongues cost shellfish"

cheers

Don

edited to add: while I was writing this a "psip login error" (408 I think) flashed up, when nothing had been done , strange??? using version 1.3pre with a frugal install of slacko 5.3 pae

[jamesbond]

Ah, it's good to let the thread ripe for a while

Some basic understanding - with this, it's easy to understand what's going on.
There is an excellent SIP tutorial if you want to read more.

I will just mention the basic points here.
1. In SIP world, in every voice call, there are 2 protocols to consider, SIP and RTP
2. SIP is used for "signalling" - that is, control the communication channel, e.g. establish a call, terminate a call, hold call etc.
3. RTP is used to actually carry the digitised voice data.

4. SIP is built from the ground-up to be interoperable.
5. Most of the SIP smarts is in the endpoint (ie in the phone) as opposed to traditional PSTN where the brain is in the telephone exchange.

6. SIP is almost peer-to-peer - call can be established directly to another endpoint, or it can be routed via a "proxy" server
that knows where the actual recipient is. In another terms, the SIP proxy server behaves like an HLR in GSM system, or like a
mail relay (or like a router).
7. RTP is peer-to-peer

Okay enough with theory. Here come some answers for the recently asked questions:

1. "Freeze problem" when closing the setup dialog - this is hardware problem. Computer is not powerful enough or
soundcard is not powerful enough to run psip. Fix - quality settings must be reduced from default 5 or 4 or 3 or 2...
As dogle has kindly explained.
Note: Change of "quality" affects the digital signal processing (DSP) load in pjsip. It has no effect on network latency etc.

2. For those who are worried about iptel.org, all you need to do is this: http://www.iptel.org/about.
An excerpt:
"This site was founded by Fraunhofer FOKUS in 1999 as part of the iptel.org research project. Fraunhofer FOKUS provides research and consulting services regarding security, VoIP services and next generation networks."

3. How about ekiga.net? A little work would reveal that (from http://wiki.ekiga.org/index.php/Ekiga.n ... bscription)
"Yes. It's free as in beer: we provide services at no charge! It's free as in speech too: Ekiga.net uses free software (SER and Asterisk). Finally, Ekiga.net's machine and internet bandwitdth is sponsored by a french company: Puce-easy9.png http://www.easyneuf.fr/"

4. Still worried? SIP servers are aplenty (SER, OpenSER, Kamalio, Yates, etc etc), all you need is a decent hosted VPS server to run them .... or you can use the peer-to-peer SIP - no need for servers.

5. I have tested psip successfully with iptel.org and ekiga.net

6. Problem with routers: pjsip.org website says that ALG causes a lot of problem for them, it's best for this to be turned off.
It's quite interesting that dpup5520 finds it otherwise.

7. users on iptel.org can call those on ekiga.net and vice versa - this is just like those on gmail.com can send email to yahoo.com.
This is true for all true SIP providers (with certain exception - e.g. the service numbers on ekiga.net (echo, callback) can only be
called when you're logged on with ekiga.net). See theory above - SIP is built for interoperability.

8. "Is it possible to hide yourself once you're logged in" - Yes, but not in the current version psip. I will do it once I find
the motivation (and the time) - meanwhile, the source is available for everybody, feel free to hack it

9. SIP protocol mandates the usage of UDP and TCP traffic, but both iptel.org and ekiga.net only supports UDP.
PSIP supports UDP, TCP and TLS (ie SSL) but the TLS support is suspect because it has never been tested - no free servers I know supports TLS.
Most free servers don't support TCP too because of capacity reasons (TCP connections require more server resources than UDP connections).
In fact, iptel.org advertises that it supports TCP when it actually doesn't - one simply lost the packet when one tries to
connect to iptel.org with TCP. Thus it's one of the troubleshooting procedure to disable TCP when you have connection problems.

10. Puppy Phone mission: It just is. Use it as you see fit, hopefully for the betterment of the world.
In other words, I don't let my hobby be guided by a "mission statement" - it significantly reduces the "fun" factor of that hobby....
anything that has a "mission statement" attached to it sounds like and smells like "work" (of which I already have plenty at the moment).

11. "secure voice call" option on the account page means: both user and caller must be using SRTP. Otherwise, psip won't establish the call.
iptel.org apparently doesn't support running SRTP - again, encrytion requires more server resources, and iptel.org being free, etc.
Note that by default, if someone calls you with SRTP - psip will use SRTP. This can be turned off in the network settings
"disable optional SRTP" - if that setting is checked, psip will reject SRTP calls.

12. Those server errors (408, 606, etc) are coming from the servers. I found that iptel.org is very fond of spitting this kind of
errors once every few days

13. Note: "once you're logged in they cannot see your IP address". Incorrect. You cannot see the IP address, but PSIP can (it just doesn't get shown in the GUI). If you turn on loglevel to 4 or above, you can see the IP address - remember the part about RTP being peer-to-peer? If PSIP does not know the IP address of the peer, it cannot establish the call.
Don't want to reveal the IP address? Use a TURN proxy - it acts a relay between you and your peer - each one will only see the IP address of the relay. But having a relay comes at a cost, *what if* the operator of that relay records every single conversation that goes through it? Hmmmm ....