This forum appears to be ignored by the "gurus"


#1

Hi guys

I have noticed that a lot of more difficult questions, mine included are never answered in this forum, yet the mailing list seems to give good responses. I wonder why? This forum is a lot easier to work with than the mailing list.

I guess I should ask this question on the mailing list. :wink:


#2

I don’t think there are any… :smile: Mostly now, I beleive that everyone that has talent is wanting to get paid for the answers to be honest… Digium is now at $175.00 per hour and many others I see are doing the same…


#3

Hmm, shame. I guess you can’t blame them. Asterisk is used a lot commercially now. I sell it myself along with Polycom phones and Cisco gateways. Still, one can live in hope :stuck_out_tongue:


#4

That seems to be a problem with a lot of the forums like this one and the one at Skype for instance. The companies don’t want to provide any support so all you have are frustrated users asking questions but no answers.


#5

There is one big problem there are very few gurus, they all have day jobs and a lot of work.

Most of them are frustrated to always have to answer the same questions over and over when a lot of those answers can be found easily with google.

They read mailinglists all the time, but dont need to come to a forum they dont have questions anyway. :smile:


#6

This leaves one in a quandry. I for instance have an integration company and we want to use Asterisk. If I want help with Cisco, I have Smartnet. If I want help with Microsoft, I use the Microsoft Partner forum.

What do I do for Asterisk? As it is open source and I am not asking a question about Digium cards, I am not sure where I can turn to for help with any questions I have.

My questions are not usually the ones that are repeated over and over in the forums. I usually spend a lot of time trying to resolve the issue myself, before resorting to user groups and forums AFTER I have searched them.

I guess, we just assume that there is always someone who has seen the problem before. I think I am dealing with a bug, or a design issue often with Asterisk, but posing questions in this forum, the developers forum and even the mailing list rarely yields replies. In fact, when I look through the forum, the questions with answers are usually very basic questions, or calls for opinions on subjects that are regurgitated over and over.

Hmm. Where to then? :confused:


#7

Mark,

There are allot of people and companies supporting Asterisk… Most like mine support systems in a certain area of the country as most installs require some on-site work…

You are correct in that with other systems you do have full support, and yes the same is here with Asterisk… If you check out Digium’s Web Site they sell a Business Edition and provide support at I believe $175.00 per hour…

Now for me that’s a bit much for Open Source, and it kinda begins to feel that this FREE OPEN SOURCE stuff is going to cost in the long run… For me nothing is FREE… when you look at REDHAT, MANDRAKE, and now SUSE they are all teaming with support plans…

I sell and support just what I feel I can handle without allot of Corportate Support, as most of my customers are very small and are looking for ways to cut dollars… I do all of this Asterisk on the side as my day job I take care of 90 million plus dollars worth of Nortel Option 81c’s Rockwell’s and Avaya G3R’s around the country…

Asterisk has a place for me in these very small 5 to 10 user systems… For me your not going to sell a $10,000.00 phone systems to a local Dentist, with just a single front desk… know what I mean… most of these small compaines are looking to reduce cost and choose open source because it’s CHEAP… like FREEE…

In responding back, like I said the compaines themselves do provide tech support, you just need to pay… I find these boards somewhat as a way to just atch what people are trying to do and sell and then I play off of it…

I make an awful lot of money with Asterisk…

Good luck… I also suggest when you do Asterisk installs use some common sense, you don’t need every feature and if you’re trying to sell them, stick to the used and needed features and get one working like you want, then just CLONE them with GHOST or something… then you never do another scratch install…


#8

I’m curious to know why so many people are moving to a system that uses UDP packets that are not reliable and will cause the audio to degrade or drop out entirely when there is much packet loss?
What are they going to tell the phone users when there is a bottleneck somewhere and calls won’t go through?


#9

I suspect at least part of the reason is because VOIP has to be realtime - or very close to it. If a packet’s delayed for more than a short time, it becomes useless.

The reliability of TCP depends on retransmission of dropped packets (which often get dropped when delays are too long). This makes certain all packets arrive at the destination - but not always in the order they were sent and sometimes quite a long time after they were originally sent. If packets were being retransmitted, the resulting audio quality would actually be worse than if they were dropped.

The retransmission itself would waste bandwidth and contribute to degradation of the resultant audio quality.

In short, it’s not the protocol that’s the problem, it’s the internet.

If the whole of the internet respected the quality of service (QoS) bits in VOIP packets, it might be a bit more useful for internet telephony - but, because of its nature, it still wouldn’t be perfect.

On the other hand, if the internet was more reliable and perfectly suited for realtime audio transmission, it would be more expensive and VOIP telephony wouldn’t be as cheap as it is.

As it is, it can be made more reliable by having more dedicated bandwidth - but a lot of that bandwidth would have to be redundant most of the time, to allow peaks in traffic to flow as quickly as possible. And bandwidth costs money.

It’s a trade-off.


#10

Interesting point Larson

I have a high end commercial IP telephony product in Zultys. Asterisk is a bit of an enigma. It is good for the 5 to 10 user site as you say, yet it can easily handle hundreds of users and multiple sites.

I do actually clone my Asterisk configs. I learnt Asterisk with AMP, but soon realised I needed to move away from AMP to get the flexibility Asterisk offers.

With regards to support, I sell it and that is my business. Because I don’t buy anything from Digium (except G.729 licences) I understand that I can’t expect free support from them, like with Zultys, who rely on me to sell their product and accordingly support me.

Asterisk is typical of the major open source projects. It is powerful, yet incomplete. It is supported by peers and in that respect well supported for basic questions, but when it gets to touch problems, the help is harder to find. I can’t really complain, because I don’t spend much time trying to help others with their problems, although I do a bit from time to time.

I will eventually crack my specific problem, which is a basic codec related issue and not a fancy extra requirement. When I do, I will support my own sites mainly with my own knowledge base. Most ongoing support will relate to my hardware, like the Cisco gateways and I can get support through Smartnet.

On the comment from SteveC5000. I must tell you that your comments here are off topic to this thread and that your ramblings show your lack of understanding of how VOIP and likely how IP works. I don’t wish to flame you, but you have mentioned on a few posts that VOIP should not use UDP and allude that all users who work with UDP are somehow backward.

The fact is that voice is approximated when it is converted to data. It does not need to be perfectly sampled and compressed. UDP is perfect for carrying this type of data. If a packet should arrive out of order, the listener would likely never know. If a packet is lost the same applies. UDP is a very low overhead protocol that is perfect for carrying voice packets. In conjunction with clever mechanisms higher up the OSI stack, like jitter buffers, the quality of voice relayed via UDP is very good. The fact is that the voice has to make it from party A to party B in a couple of hundred milliseconds MAX! All the state information and packet reassembly in the world cannot make the data get there any quicker.

That is why UDP is used.

Cheers all

Mark


#11

[quote=“stevec5000”]I’m curious to know why so many people are moving to a system that uses UDP packets that are not reliable and will cause the audio to degrade or drop out entirely when there is much packet loss?
What are they going to tell the phone users when there is a bottleneck somewhere and calls won’t go through?[/quote]

tcp would work fine, if you want the users to hear the beginning of the sentence after the end, since voice is real time it won’t tolerate out of order packets. dead spots in a conversation sound better than garbled voices


#12

[quote=“stevec5000”]I’m curious to know why so many people are moving to a system that uses UDP packets that are not reliable and will cause the audio to degrade or drop out entirely when there is much packet loss?
What are they going to tell the phone users when there is a bottleneck somewhere and calls won’t go through?[/quote]

This is where QOS and good design begin. Then all you have to worry about is the medium. Since most people have access to broadband, an up-to 128K stream isn’t that big of a deal, as long as you’re not downloading mpegs at the same time or similar. Then if you know how to configure an access list on a router you can even make it where that won’t matter.

At the present generation of codecs and growing stability of high-speed backbones, at G711, IP calls can achieve a MOS 5 rating, which means most people can’t tell the difference between VoIP and a circuit-switched toll call.

So then it becomes a matter of media failure, which makes us tell the customer the same thing a LEC or CLEC has to tell them when their media fails, which is ‘dude, we need to troubleshoot. we got errors on your ____ (LAN, T1, T3, OCx)’

Join us in the new millenium.


#13

Additionally, most (good) VoIP servers & clients send staggered, MULTIPLE copies of the same UDP packet with the same sequence number, and the receiver puts them in order and throws the excess away.


#14

You’ve got to be kidding!?

Sending multiple copies of the UDP packets would double/triple/quadruple, whatever, the bandwidth used for the call. That would be more likely to degrade the sound quality, rather than improve it, over a path with limited bandwidth.


#15

Mark:

The problem is being fixed as we speak, but not as fast as we all would like.

Asterisk is simply not mature enough. There’s lots and lots of people jumping into the Asterisk bandwagon, but a this stage there are many more questions than answers. I have seen it dozens of times: the IBM PC marketplace back in 1981 was like this. People formed “user groups” because there was no Internet. Years later, the biggest group of user groups in the world, the Boston Computer Society, closed its doors because it was no need for it anymore.

But the best example is Cisco (or even Oracle): there is space for commercial support, for free support, tons of books, video training, etc., etc.

In time, Asterisk will approach that level of maturity and have a wide variety of support choices.

In the meantime, you should be satisfied to say in the future that you were one of the Asterisk pioneers.

Having said that, I agree in that the forum is so much more convenient than a mailing list. In fact, I wish we had an Asterisk Usenet newsgroup.
That will be a milestone of marketplace maturity!

-RFH


#16

I second that.

Asterisk is alone here. I am a computer consultant, and yet I have an open-source company (NeoSmart Technologies at NeoSmart.net/) and i spend my free time on support forums like ProNet, NeoWin, ieXbeta, and my own.


#17

[quote=“twohig5”]
tcp would work fine, if you want the users to hear the beginning of the sentence after the end, since voice is real time it won’t tolerate out of order packets. dead spots in a conversation sound better than garbled voices[/quote]

Where does the myth come from that packets are delivered to the application out-of-order???

With tcp you get either the correct data, or no data. If a packet gets lost all following received packets are not delivered, but buffered up.
When the missing packet is finally retransmitted, it and all the queued packets are delivered to the application in order. Maybe a second, or two too late, bit every bit is correct.

That is very noticeable, whereas 20ms missing from the audio stream is not.

The reason there is tcp and udp is that both have an application.
For VoIP udp is the correct choice.