Puppy Linux Discussion Forum

[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]

[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.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

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 ....

[nooby]

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

[Stripe]

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 )

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

[nooby]

Is it a "Sunday conference" now tomorrow maybe?

[gcmartin]

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.

[smokey01]

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

[gcmartin]

[Lobster]

[nooby]

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.

[nooby]

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?

[nooby]

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