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Puppy Phone - VOIP using SIP
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Stripe

Joined: 23 Jun 2010
Posts: 658
Location: In a field. England

PostPosted: Thu 03 Nov 2011, 14:08    Post_subject:  

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 Wink )

Cheers

Don

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gcmartin

Joined: 14 Oct 2005
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Location: Earth

PostPosted: Thu 03 Nov 2011, 14:41    Post_subject:  

Stripe wrote:
..
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.

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gcmartin

Joined: 14 Oct 2005
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PostPosted: Thu 03 Nov 2011, 14:55    Post_subject:  

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.

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Stripe

Joined: 23 Jun 2010
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Location: In a field. England

PostPosted: Thu 03 Nov 2011, 15:10    Post_subject:  

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" Wink

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

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jamesbond

Joined: 26 Feb 2007
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PostPosted: Fri 04 Nov 2011, 07:24    Post_subject:  

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

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.net_VoIP_service_subscription)
"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 Smile

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 Smile

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 ....

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nooby

Joined: 29 Jun 2008
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Location: SwedenEurope

PostPosted: Fri 04 Nov 2011, 09:54    Post_subject:  

Thanks James, most likely I will try to be active coming Sunday.
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Stripe

Joined: 23 Jun 2010
Posts: 658
Location: In a field. England

PostPosted: Fri 04 Nov 2011, 14:01    Post_subject:  

Hi all

@jamesbond Thanks for the info it is very useful and helpful in understanding the protocols and how they work (well for me anyway Embarassed )

RE: dropping out, CatDude and I are currently working our way through the previous source files to try and identify a time when the dropping out became a problem (your timeline is brilliant for this) and we will hopefully be able to pinpoint when the dropping out first became noticeable/a problem. so far we have it some where between [add93b0db4] 2011-10-05 (02:06) and [589f28944b] 2011-10-01 (14:49) but will post our full findings if we find anything conclusive.
any guidance/advice gladly taken

thanks again James/everybody for a brilliant tool and hope this helps

Don
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nooby

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PostPosted: Sat 05 Nov 2011, 16:35    Post_subject:  

Is it a "Sunday conference" now tomorrow maybe? Smile
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gcmartin

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PostPosted: Sat 05 Nov 2011, 20:52    Post_subject:  

Thanks @Stripe, @Jamesbond, and everyone who contributes here.

I have a single additional comment and I have 2 questions (that I think scares people when I ask)

I want to add one thing that JamesBond has already commented on, but, I want to say it a little differently.
Quote:
When a browser sends out your request to connect to Yahoo.com, something in the Internet MUST convert that so that your browser can "talk" to the Yahoo site. That something is a DNS who translates and tells your browser "where" Yahoo is. (Oh you could just enter the IP address but most of us don't walk around with a list of them in our pockets for use)

In the SIP world, when you want to connect (talk) to someone, there MUST be someone-somewhere who can do that for your SIP device (PSIP/PuppyPhone/etc.) For us PC users, that is typically a Registrar like IPtel/Ekiga/etc. (Actually, a Registrar does a wee-bit more)

Thus, a Registrar is for us, SIP phone users, like a DNS is for our browser use.
2 Questions
  1. Has any member of this thread-forum used PuppyPhone to "connect" to a "non-PC client" SIP device? (There are plenty on SIP devices that don't need PCs for calling)
  2. Has any member used his smartphone or blue-tooth headset to handle his PuppyPhone calls?
Thanks in advance for any help on the questions

And, Thanks, again, to Smokey01 and JamesBond and Lobster and Nooby and DPUP5520 and .... Everyone! for this contribution to Puppyland

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smokey01


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PostPosted: Sat 05 Nov 2011, 21:28    Post_subject:  

gcmartin, jamesbond has, here:
http://www.murga-linux.com/puppy/viewtopic.php?p=565609#565609

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gcmartin

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PostPosted: Sat 05 Nov 2011, 22:31    Post_subject:  

smokey01 wrote:
gcmartin, jamesbond has, here:
http://www.murga-linux.com/puppy/viewtopic.php?p=565609#565609
Thanks Smokey01! That's great news. He's using his Nokia N900 with its SIP client on the Nokia OS to connect to PSIP. This is a shining example of using this product to connect to a SIP device.

2 Questions asked a little differently
  • @JamesBond. Did you do this via the cell network or via Wifi over your WLAN for calls to your "SIP device"? (I know it should work the same.)
  • Has any member used a secondary PC 'heaset' device (USB webcam/blue-tooth headset/2nd soundcard) to converse using his PuppyPhone?
Thanks for your response, @Smokey01. I missed that section of this thread.

Edit (110611) - Thanks DPUP5520. We now have confirmation of use of the SIP client on the cell Phone in connecting into conference room services using Cell network feature of the Phone.

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Lobster
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PostPosted: Sat 05 Nov 2011, 22:34    Post_subject:  

nooby wrote:
Is it a "Sunday conference" now tomorrow maybe? Smile


People are now logging into the conference call during the week too . . .
**Conference call on Sundays at 1200h GMT** Next Sun 6 Nov
From the menu item Iptel, click on conference to enter - our number is 528

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nooby

Joined: 29 Jun 2008
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PostPosted: Sun 06 Nov 2011, 02:31    Post_subject:  

Thanks Lobster, most likely I will try to be a bit active too.

I still have problem deciding on levels on the Retrovol.

My earphone sound is too low. I barely hear what people say.
So I will try to get it through an outer amplifier into the ear phones
but to take the microphone through the ordinary jack socket.

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nooby

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PostPosted: Sun 06 Nov 2011, 02:35    Post_subject:  

Re reliability of SIP services.

I know too little but this is maybe something else then
not comparable to the cost of having iptel going?

Quote:
"We are sorry to inform you that due to the rising cost of operations, we have been forced to discontinue the Voxalot VoIP service. This change will be effective from December 31st, 2011.


So that was a VoIP and and Iptel only provide the IP DNS something?
Our IP is translated into a nick name? So them don't really have a VoIP
which cost a lot of money to have going? [/url]

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nooby

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PostPosted: Sun 06 Nov 2011, 08:08    Post_subject:  

I can not hear a single person. is it really room 528?
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