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VoipTunnel Edit

I set the box to Lock to Gateway and bingo >everything works. However, the box (DN01) was set to 'Any' and it >was working perfectly. So far being behind a NAT does not seem to be >an issue as I have always had set "Gateway NAT workaround". Say I >have a repeater that is at the end of a 5 hop mesh and I set "Lock >to Gateway" where the Gateway is 5 hops away. Does the signal beam >directly to the Gateway or does it utilize the other repeaters and >create a mesh as it is designed to do.

It utilised the other nodes (as in creating a mesh), but the VOIP traffic goes over the tunnel, so it doesn't matter what other nodes are in between.

* VoipPorts
* SkyPe

GatewayAndvoip Edit

{{{ Thanks for your help. I set the box to ‘Lock to Gateway’ and bingo everything works. However, the box (DN01) was set to ‘Any’ and it was working perfectly. So far being behind a NAT does not seem to be an issue as I have always had set “Gateway NAT workaround”. Say I have a repeater that is at the end of a 5 hop mesh and I set “Lock to Gateway” where the Gateway is 5 hops away. Does the signal beam directly to the Gateway or does it utilize the other repeaters and create a mesh as it is designed to do.


Set "Lock to Gateway" to your gateway node on DN01,DN02. Also looks like your gateway is behind a NAT, try "Gateway NAT workaround" setting on GW. I didn't get VoIP to work until we put our GW on a public IP.


I have been having issues with my VOIP setup on one of my 2 Downlink Nodes. The VOIP phone connects and registers on my Gateway Node (GN) and my first Downlink Node (DN01), however I can’t register on my second Downlink Node (DN02). My setup is as follows:


Gateway Node (GN)

Box function: Always Gateway

Cross mesh routing: StandardIA

Gateway use DHCP dns: No

Daisy chain gateway's dns: Yes

static eth addr: 192.168.1.5

static eth netmask: 255.255.255.0

static eth gateway: 192.168.1.1

static eth dns: 192.168.1.1

Wired local 192.168.: 1.5


My Gateway node is a wired connection from my router and has been given a static address as outlined above.


DN01

Box function: Auto Detect Cross mesh routing: Standard IA Gateway use DHCP dns: No Daisy chain gateway's dns: Yes static eth addr: static eth netmask: static eth gateway: static eth dns: Wired local 192.168.: 2.5 Lock to gateway: Any At the moment DN01 has no wired clients and works very well with the VOIP phone.


DN02 Box function: Auto Detect Cross mesh routing: Standard IA Gateway use DHCP dns: No Daisy chain gateway's dns: Yes static eth addr: static eth netmask: static eth gateway: static eth dns: Wired local 192.168.: 3.5 Lock to gateway: Any The only difference between DN01 and DN02 is the wired local client field. As with DN01, DN02 does not have any wired clients. Can anyone see an obvious problem with the configuration of DN02. I have tried everything to connect to client DN02 with the VOIP phone without any luck. I have uninstalled and reinstalled the SIP proxy. All boxes are running on build 25dev87 and all have been working with the internet and email.

}}}


preferred gateway Edit

Unfortunately you can not have your cake and eat it ;) It depends how you have implemented VOIP, if you have used the LW SIP Proxy then I believe you are correct, if you use an external SIP provider with good NAT support then usually no, if you use IAX with Asterisk then absolutely not as IAX just works :)

===Edit

Thanks a lot for your answer to my question. If I remove the "lock to gateway" option would VoIP continue to function. As far as I know in order for VoIP to work the nodes need to be locked to gateway.

> Lock to gateway is exactly that, it locks the node to the > gateway chosen in the drop down list, if you want it to swap > don't lock it, set lock to gateway to "ANY" and then set your > preferred and 2nd preferred gateways to the ones you want to > use but don't forget that it will only swap if the preferred > gateway goes down and not if the ADSL link dies. You can > test for a failed link but it does not do this by default. >

=========Edit

> I have two GW nodes, 2XADSL. Today one GW was down for more > than 3 hours. The nodes connected to this GW which is > Preferred gateway for these nodes DID NOT switch to 2nd > Preferred gateway. In wiana they are all set "lock to > gateway". Does "lock to gateway" will prevent the nodes to > use 2nd Preferred gateway.

=Edit

> I want to change the GW node with more powerful machine. What > would change if I just remove the CF card from the old > machine and put it into the new machine. Does the HDW Key > will change as well.

==Edit

[[SipProxy

TableOfContents

Er, what does "ps | grep sip" do? I typed it in after ssh'ing into the box and nothing happened (or perhaps it did - it's just that there wasn't any on-screen feedback). The sipproxy module is shown as off on the software module manager, I guess you are suggesting that even though it isn't showing it is infact loaded.

Are you sure it's not installed?

Check using

. ps | grep sip

If its is, and you doon't want it to be, make sure you toggle it in the Software Module manager and then run 'swman' on then node.


SIPPROXY

Hello All Some time back I was having a problem with my VoIP connections and whilst trying to resolve the problem I removed the sipproxy module from all of my nodes.

After a bit of mucking about I got the phones to work but I never put sipproxy back on. 4 weeks or so later all is working well on the VoIP front and still no sipproxy.

I'm confused, I thought that VoIP shouldn't work without the module installed. I'm running dev98.

Hello All Some time back I was having a problem with my VoIP connections and whilst trying to resolve the problem I removed the sipproxy module from all of my nodes.

After a bit of mucking about I got the phones to work but I never put sipproxy back on. 4 weeks or so later all is working well on the VoIP front and still no sipproxy.

I'm confused, I thought that VoIP shouldn't work without the module installed. I'm running dev98.

If you're using a sip provider with nat traversal (typically by setting the outbound sip proxy option when configuring your sip device) then it is likely that you don't need the sipproxy module to make and receive calls.

The downside of this approach is that intra-mesh calls will go out of your gateway and back in again rather than being routed internally on the mesh.



Some time back I was having a problem with my VoIP connections and > whilst trying to resolve the problem I removed the sipproxy module from > all of my nodes. > > After a bit of mucking about I got the phones to work but I never put > sipproxy back on. 4 weeks or so later all is working well on the VoIP > front and still no sipproxy. > > I'm confused, I thought that VoIP shouldn't work without the module > installed. I'm running dev98.

asdfasf Edit

http://www.lemontreenetwork.com/probs/486config.htm We are running de-centralized Asterisk servers on our networks (i.e. on each mesh node), to which the SIP-clients are registered. The Asterisks are then setup to choose the cheapest and most efficient route/provider for outbound calls (i.e. off mesh) using IAX2 trunking instead of SIP. IAX2 is much more powerful and forgiving than SIP when it comes to complicated network infrastructures like Mesh. Plus, the mesh-sipproxy implementation does currently not work on dual-radio nodes if your clients are on wlan1 interface.

We noticed that in the Asterisk sip.conf file all users needed to have "canreinvite=yes" to make the intra-mesh calls work with the sipproxy. Maybe your provider want to keep track of all calls and have this setting to "no"?

Go for asterisk! Just a thought :)



It's been a week since I last posted the message with the SIPPROXY issue and my thanks to everyone for their help. I want the benefits that Jon describes - intra-mesh calls are quite popular especially with my wife :) In summary I have taken sipproxy off, loaded it again, updated one node to dev100 taken sipproxy off then loaded it again, completely re-installed dev98 on another node - all to no avail. If sipproxy goes on my phones die. My clients are a patient bunch and realise that I am doing my best - they don't understand that it is a technical issue that I am grappling with in that I don't want to use up ADSL bandwidth for intra-mesh calls. Currently then sipproxy is off, so I don't get the flashing red light on my grandstream 286's & 486's and the phones are working. I know my sip provider installed a new sip server (asterisk) a few weeks ago, by any chance could the problem be their end? I have stuck my Grandstream configuration page on the web at http://www.lemontreenetwork.com/probs/486config.htm Maybe I have set the phone up incorrectly, but considering it has been working for 6 months, well, who knows?

> wasn't any on-screen feedback). The sipproxy module is shown as off on > the software module manager, I guess you are suggesting that even > though it isn't showing it is infact loaded. > > Rgds > Mike > >

SIPPROXY > > > Hello All > Some time back I was having a problem with my VoIP connections and > whilst trying to resolve the problem I removed the sipproxy module > from all of my nodes. > > After a bit of mucking about I got the phones to work but I never put > sipproxy back on. 4 weeks or so later all is working well on the VoIP

> front and still no sipproxy. > > I'm confused, I thought that VoIP shouldn't work without the module > installed. I'm running dev98. > > Any ideas anyone? > Rgds > Mike >


It's been a week since I last posted the message with the SIPPROXY issue and my thanks to everyone for their help.

I want the benefits that Jon describes - intra-mesh calls are quite popular especially with my wife :)

In summary I have taken sipproxy off, loaded it again, updated one node to dev100 taken sipproxy off then loaded it again, completely re-installed dev98 on another node - all to no avail.

If sipproxy goes on my phones die. My clients are a patient bunch and realise that I am doing my best - they don't understand that it is a technical issue that I am grappling with in that I don't want to use up ADSL bandwidth for intra-mesh calls.

Currently then sipproxy is off, so I don't get the flashing red light on my grandstream 286's & 486's and the phones are working.

I know my sip provider installed a new sip server (asterisk) a few weeks ago, by any chance could the problem be their end?

I have stuck my Grandstream configuration page on the web at http://www.lemontreenetwork.com/probs/486config.htm Maybe I have set the phone up incorrectly, but considering it has been working for 6 months, well, who knows?

Er, what does "ps | grep sip" do?  I typed it in after ssh'ing into > the box and nothing happened (or perhaps it did - it's just that  there

> wasn't any on-screen feedback). The sipproxy module is shown as off on

> the software module manager, I guess you are suggesting that even > though it isn't showing it is infact loaded. > > Rgds > Mike > >



HopsAndvoip Edit

>Thanks for your help. I set the box to Lock to Gateway and bingo >everything works. However, the box (DN01) was set to 'Any' and it >was working perfectly. So far being behind a NAT does not seem to be >an issue as I have always had set Gateway NAT workaround. Say I >have a RepeaTer that is at the end of a 5 hop mesh and I set "Lock >to Gateway" where the Gateway is 5 hops away. Does the signal beam >directly to the Gateway or does it utilize the other repeaters and >create a mesh as it is designed to do.

It utilised the other nodes (as in creating a mesh), but the VOIP traffic goes over the tunnel, so it doesn't matter what other nodes are in between.


Can you please give me some more info about adding DNS address to the node in order for VoIP to work. What DNS address needs to be added in wiana? Is it for GW node only or to all nodes? I had difficult time to get Grandstream 286 to work. I had to uninstall and reinstall sip proxy few times to GW node and the node serving the 286 with DHCP few times in order to get it working. I had "lock to gateway" enabled in wiana.


Ok we had this some time back, I think we solved it by adding a static DNS address to the node. I trust you have port forwarded all to relevant ports sip udp 5060, RTP ports.

Let us know! if this solves it, but sure this is what we did. I am trying to get a Grandstream handytone 486 to work thru a gateway node on my test network. I have gone thru the setup in the wiki and also looked thru old mailing list posts but no luck.

I am using a wrt54g to connect to my mesh gateway. The wrt is set to get ip dhcp from the gateway and I am putting out dhcp to the grandstream. The wrt setup works on another non mesh ap I have here that goes to the same provider, so I must be missing something in the mesh gateway setup. Any help would be greatly appreciated. I am usig dev106 on this gateway.


* SkyPe

AsterIsk Edit

TableOfContents

Load the SIP proxy from wiana. We have users running sip on or mesh network and they have no problems, remember to open your sip ports on the router for external traffic.

I have xlite sip phone on a windows xp desktop connected via LAN to an sme server 6.0.1 on which i have compiled asterisk. I can make a call and hear the demo config. Also onnected to the LAN is a single meshAP. What do i need to do to get xlite running on my laptop via the mesh ? At the moment i can see it attempting to connect but packets sent back dont seem to get through.

>The last option is the 'Gateway NAT Workarounds', can anyone explain >just what this does, I don't remember seeing it mentioned before on >the mailing list or in any docs.

This allows workarounds for VOIP to work where there is further NAT at the gateway. The issue is to do with the fact that the VOIP gateway (meshbox) has to advertise the relevant public IP address.

I would seriously take a look at IceCast http://www.icecast.org/ I havent experimented with it on our mesh yet but it does appear to meet most of the requirements.

There is a development version of an asterisk software module which can be loaded onto a node. This can be used in various ways to perform useful voice over ip functions, but needs some additional configuration to become more flexible.

To load the software module, select "software module manager" from the advanced menu on the node management page and select the "Asterisk" module option. Please note that currently the asterisk module is not compatible with the sipproxy module and so if you are installing the asterisk module then the sipproxy should be unchecked! This is because at the moment they would both try to use the same sip port number.

You can run swman, or your node may automatically update and asterisk will be installed in /inst/asterisk and automatically started.

To prototype configurations with the MeshAP asterisk server you should first be familiar with asterisk functionality. Please see www.asterisk.org for

further information.

The development MeshAP asterisk server is a standard install of Asterisk-1.0.0 - The only functionality removed is that of mp3 music on hold. Conferencing is not yet included as this requires use of a special hardware timer.

The configuration files are stored in /inst/asterisk/etc - New agi files can be placed in /inst/asterisk/extra-agi-bin - New sound files can be placed in /inst/asterisk/extrasounds

To access the debug console use /inst/asterisk/ro/bin/asterisk -r

Try making IAX or SIP calls to the IP address of the node, you will be greeted with the standard asterisk demo configuration. Extension 400 is a speaking clock, extension 500 calls digium, 600 is an echo test, dialing 9 followed by an international number in european format will make an outgoing call via the locustworld switch. For example 90012125551212 would make a call to a directory assistance number in the USA.

Please note that logs and voicemails are stored on the compact flash media in the default configuration and so care should be taken not to exceed the available space. The MeshAP asterisk module fits into approx 2mb as standard.

Users are strongly encouraged to discuss useful configurations on the voiplive mailing list and to submit dialplans and configuration files so that the module can be utilized for many different voip applications and setups. LocustWorld are working on this but welcome participation and feedback.

Hints:

To use IAX you set your iax host to the ip address of the node and your username to guest and password to nothing. Then simply dial extensions. If you don't already have the iaxcomm package you can get it from http://iaxclient.sourceforge.net/iaxcomm/ or by mailing

Wiki page: http://locustworld.com/tracker/wiki?p=AsteriskSoftwareModule

featerues Edit

In the settings of the SIP phone, usally where you set the SIP-server address. Could also be a buggy phone. Which ones is it?

The mesh gateway has a static IP address and so does the SIP phone. Where is the Register Expiration time set?

Try to use a fixed IP-adress, and/or shorten the Register Expiration time (usually 3600 secs), at the SIP client/phone. /Stefan

I am having major hassles with my Voip phone running on the mesh. The system is fine for a couple of days when seemingly with no reason (no config changes have been make and the meshbox has been up for 8-9 days) I can not receive incoming calls. This is very frustrating as I am missing some important calls - the caller receives a busy signal. Could this have something to do with DHCP leases? I have installed the sipproxy and am running on build25dev87. Any insight would be appreciated.

> > btw - is anyone using an Asterisk/LW set-up? > Yep, we are. I had a feeling you might be ;)

> VoIP is the "next big thing" on internet, and VoIP over Mesh is *the* killer-app! > Telcos beware! ;) Totally Agree.

My prime application for my soon to be started (hopefully) WISP is VoIP as none of my clients have a 'phone line installed (can't get one) and calling home to Germany/UK/Netherlands/Canada on a mobile is starting to hurt their respective pockets. The only other app. they are interested in is email - no films/music/surfing, none of that so I'm really keen to sqeeze as much as I can out of my "rip-off 92 euros a month" 128/512 ADSL connection.

I understand from a recent posting by Jon that IP phones will be able to interface with other IP phones on a LW network just by dialling the IP address. This is great, anyone doing it?

Presumably adding Asterisk to Jon's scenario increases functionality (e.g. voice mail etc). I would want to stay away from adding my own telephone lines to an Asterisk server (I'm not interested in supplying a full blown pbx) but would like to ensure some QoS for the link to the Internet and supply some of the goodies to my prospective clients. Stefan, is it really an out of the box/standard solution as you suggest or are you really a bit of a Linux/telecomms wizard ('cos I ain't)?


> > btw - is anyone using an Asterisk/LW set-up? > Yep, we are. And it works great over the Mesh even without the "not so > decent codecs" ;) > It's quite easy to control QoS inside your own net, but when you hit the > wild web, things start to get out of control. > One of our customers are using Voip from Sweden to China, standard > (free)Asterisk setup and 4 Meshhops then out over internet to their > local office in Shanghai, and 95% of the calls are "better" quality (64 > kbit/s compared to PSTNs 8 kbit/s) than long distance PSTN (which > usually these days is a mix of VoIP/PSTN, with all local telcos cutting > into the market), sometimes they get a "clicky" sound, like a bad GSM > connection, but still OK. > Since they have reduced their monthly telephone bills with 90% also, > they're not complaining! > > VoIP is the "next big thing" on internet, and VoIP over Mesh is *the* > killer-app!

> > Do you have any QoS or bandwidth control running so that your voip > > traffic > is prioritised. > > You tend to find that Voip likes consistant, steady badnwidth and ping > times. When you > > say 128K do you mean at the broadband end in which case is it a > > contended > service? > > i.e. are you sure you're getting the uplink speed you think you are? > > No QoS/bandwidth control in place, but of course "am I getting the speed > I think I am" is valid since I use ADSL as backhaul. > > > Using as decent codec like G.729 or G.723 your bandwidth demand should > > > be > around > > 24K (bits per second). > Thanks, I'll try G.723 with a frames/packet setting of 8. > For info, I found a faq for the Grandstream BudgeTone 100 - > http://www.grandstream.com/FAQ.htm. I can see that using the factory > settings my bandwidth would have been chewed up since the default is for > "toll" quality (64k).

btw - is anyone using an Asterisk/LW set-up?

> I shall have a play - thanks all

> > Using as decent codec like G.729 or G.723 your bandwidth demand should > > > be > around > > 24K (bits per second). If you're using some of the cra**ier 'free' > > codecs > like G.711 then > > this could be double or tripple. If your voip equipment has parameters > > > to > control the > > frames/packet setting then try increasing this (upto 8 is usually > > okay). > This will > > significantly reduce your badnwidth demand. > > > > Do you have any QoS or bandwidth control running so that your voip > > traffic > is prioritised. > > You tend to find that Voip likes consistant, steady badnwidth and ping > times. When you > > say 128K do you mean at the broadband end in which case is it a > > contended > service? > > i.e. are you sure you're getting the uplink speed you think you are? > > > > On 26 Jul 2004 at 12:59, Mike Crompton wrote: > > > > > Hello All > > > I've been reading the recent questions regading VoIP with some > > > interest. The following may have been covered previously, if so > > > apologies. > > > > > > My testing is showing that the bandwidth required for my SIP phone > > > is greater than I am expecting it to be. I am not looking for > > > "studio" quality voice, in fact the complete opposite. A "tinny" > > > sounding conversation would be good enough for me. What I do need > > > is for the conversation to be uninterrupted. I figure that the > > > lower quality the sound the less bandwidth required - is this > > > assumption correct? > > > > > > The problem is that conversations I have are "broken up" unless I > > > make around 128k available for the upstream link - this just doesn't > > > > agree with the stuff I have read where I am told that 32k should be > > > enough, again bearing in mind with the quality I am prepared to live > > > > with. > > > > > > In my testing, I have ended up taking the wireless element out of > > > the equation (why complicate things further!), instead just hooking > > > my 'phone into my wired LAN. WAN access is via ADSL. > > > > > > So I am playing around with the various codecs as shown on my > > > Grandstream 'phone. Am I on the right track and if so does anyone > > > know which codec I should be using? > > >

==Edit

One of the cool features with Asterisk is that it can act as an client (phone) to your provider, and as acting as server to your "real" phones, i.e. it can translate between different codecs. It helps if you know some Linux, but there are many good "how-tos" on setting up an Asterisk box. And here I would normally say:"And we would gladly help you set one up for $100/hour", as the consultant I am. ;)

But what a heck... Like in the saying:” If one butterfly starts flapping it's wings, it can (will?!!) start an earthquake on the other side of earth", so let's start flapping our wings shall we!? Anyway, who needs allot of paper with of $'s or ˆ's on them?? It's still just a piece of paper, created without backing of any real value! Real value comes out of PEOPLE HELPING EACH OTHER reaching higher grounds. Greed and profit-hunger will be mankind’s doom. Time to change anytime soon are we? Clock's ticking..

Take a PC- PII300+, 128Mb+ RAM and 10GB+ HD with Linux compatible hardware, i.e. NIC: Follow this: 1. http://www.projektfarm.com/en/support/debian_setup/index.html You may skip the web/mail/dns/ftp/pop3/webalizer parts (keep mySQL if you want to do accounting/logging etc) 2. When your box is up and running: http://www.azxws.com/asterisk/asterisk-debian-howto.html You can skip everything that belongs to Zaptel setup as you don't want PSTN termination on your box. (Who will need PSTN in the future!? ;) 3. Then scroll down to "Setting up SIP" on : http://www.automated.it/guidetoasterisk.htm (Or follow this from top if you like Red Hat flavour) . If you ask me, go for Debian! 4. Or try the Asterisk Live! CD :http://www.automated.it/asterisk/ (No tried it myself, let me know if you do, and how it works?!)

That shall get you going, drop me a mail when you get stuck and I'll try to help you if I can.

P.S. Look, this is not an open invitation to bombard me with mail, please folks!? True knowledge comes by doing it yourself (Often the hard way) End of today's preaching.. ;)

> > btw - is anyone using an Asterisk/LW set-up? > Yep, we are. I had a feeling you might be ;)

> VoIP is the "next big thing" on internet, and VoIP over Mesh is *the* killer-app! > Telcos beware! ;) Totally Agree.

My prime application for my soon to be started (hopefully) WISP is VoIP as none of my clients have a 'phone line installed (can't get one) and calling home to Germany/UK/Netherlands/Canada on a mobile is starting to hurt their respective pockets. The only other app. they are interested in is email - no films/music/surfing, none of that so I'm really keen to sqeeze as much as I can out of my "rip-off 92 euros a month" 128/512 ADSL connection.

I understand from a recent posting by Jon that IP phones will be able to interface with other IP phones on a LW network just by dialling the IP address. This is great, anyone doing it?

Presumably adding Asterisk to Jon's scenario increases functionality (e.g. voice mail etc). I would want to stay away from adding my own telephone lines to an Asterisk server (I'm not interested in supplying a full blown pbx) but would like to ensure some QoS for the link to the Internet and supply some of the goodies to my prospective clients. Stefan, is it really an out of the box/standard solution as you suggest or are you really a bit of a Linux/telecomms wizard ('cos I ain't)?

> > btw - is anyone using an Asterisk/LW set-up? > Yep, we are. And it works great over the Mesh even without the "not so

> decent codecs" ;) It's quite easy to control QoS inside your own net, > but when you hit the wild web, things start to get out of control. > One of our customers are using Voip from Sweden to China, standard > (free)Asterisk setup and 4 Meshhops then out over internet to their > local office in Shanghai, and 95% of the calls are "better" quality (64 > kbit/s compared to PSTNs 8 kbit/s) than long distance PSTN (which > usually these days is a mix of VoIP/PSTN, with all local telcos cutting > into the market), sometimes they get a "clicky" sound, like a bad GSM > connection, but still OK. > Since they have reduced their monthly telephone bills with 90% also, > they're not complaining! > > VoIP is the "next big thing" on internet, and VoIP over Mesh is *the* > killer-app! Telcos beware! ;)

> > Do you have any QoS or bandwidth control running so that your voip > > traffic > is prioritised. > > You tend to find that Voip likes consistant, steady badnwidth and > > ping > times. When you > > say 128K do you mean at the broadband end in which case is it a > > contended > service? > > i.e. are you sure you're getting the uplink speed you think you are?

> No QoS/bandwidth control in place, but of course "am I getting the > speed I think I am" is valid since I use ADSL as backhaul.

> > Using as decent codec like G.729 or G.723 your bandwidth demand > > should > > > be > around > > 24K (bits per second). > Thanks, I'll try G.723 with a frames/packet setting of 8. > For info, I found a faq for the Grandstream BudgeTone 100 - > http://www.grandstream.com/FAQ.htm. I can see that using the factory > settings my bandwidth would have been chewed up since the default is > for "toll" quality (64k). > > btw - is anyone using an Asterisk/LW set-up? > > I shall have a play - thanks all

> > Using as decent codec like G.729 or G.723 your bandwidth demand > > should

> > be > around > > 24K (bits per second). If you're using some of the cra**ier 'free' > > codecs > like G.711 then > > this could be double or tripple. If your voip equipment has > > parameters

> > to > control the > > frames/packet setting then try increasing this (upto 8 is usually > > okay). > This will > > significantly reduce your badnwidth demand. > > > > Do you have any QoS or bandwidth control running so that your voip > > traffic > is prioritised. > > You tend to find that Voip likes consistant, steady badnwidth and > > ping > times. When you > > say 128K do you mean at the broadband end in which case is it a > > contended > service? > > i.e. are you sure you're getting the uplink speed you think you are?

> > > I've been reading the recent questions regading VoIP with some > > > interest. The following may have been covered previously, if so > > > apologies. > > > > > > My testing is showing that the bandwidth required for my SIP phone

> > > is greater than I am expecting it to be. I am not looking for > > > "studio" quality voice, in fact the complete opposite. A "tinny" > > > sounding conversation would be good enough for me. What I do need

> > > is for the conversation to be uninterrupted. I figure that the > > > lower quality the sound the less bandwidth required - is this > > > assumption correct? > > > > > > The problem is that conversations I have are "broken up" unless I > > > make around 128k available for the upstream link - this just > > > doesn't > > > > agree with the stuff I have read where I am told that 32k should > > > be enough, again bearing in mind with the quality I am prepared to

> > > live > > > > with. > > > > > > In my testing, I have ended up taking the wireless element out of > > > the equation (why complicate things further!), instead just > > > hooking my 'phone into my wired LAN. WAN access is via ADSL. > > > > > > So I am playing around with the various codecs as shown on my > > > Grandstream 'phone. Am I on the right track and if so does anyone > > > know which codec I should be using?


Ehh, sorry my brain's abit overheated, should have been smoothwall http://smoothwall.org/  ;) You could also set up a Asterisk box on a spare PC, then you can illiminate all "outside" factors. http://www.asterisk.org/ It's not so hard if you only want to experiment with SIP, and gives you alot of other col functions, like voicemail, conference etc.

   >> Uncompressed usually requires around 64 Kbit/s for each channel, that’s 128 kbit/s for full duplex.
    
   I didn't know that, thanks.  It also seems to explain why I need to allocate 128k, I will check with my SIP provider.
    
   Regarding which codec could be used, I wasn't expecting to have to include my SIP provider in the "testing equation", you've just saved me a heck of a lot of useless testing by the sounds of it, much appreciated. :)
    

Which Codec you can use depends on what your SIP provider can offer, if they support GSM, than you can get away with around 8 kbit/s with quite decent quality. Uncompressed usually requires around 64 Kbit/s for each channel, that’s 128 kbit/s for full duplex. ADSL is asymmetric (lower speed up than down) and you are sharing the providers bandwidth with all other ADSL users on your exchange. If you are serious about VoIP then try to get a dedicated (T1) line, if you can (and can afford), and route all SIP traffic down this line. Or atleast try a QoS aware router between the mesh and your uplink (RouterOS or moonwall) I've been reading the recent questions regading VoIP with some interest. The following may have been covered previously, if so apologies. My testing is showing that the bandwidth required for my SIP phone is greater than I am expecting it to be. I am not looking for "studio" quality voice, in fact the complete opposite. A "tinny" sounding conversation would be good enough for me. What I do need is for the conversation to be uninterrupted. I figure that the lower quality the sound the less bandwidth required - is this assumption correct? The problem is that conversations I have are "broken up" unless I make around 128k available for the upstream link - this just doesn't agree with the stuff I have read where I am told that 32k should be enough, again bearing in mind with the quality I am prepared to live with. In my testing, I have ended up taking the wireless element out of the equation (why complicate things further!), instead just hooking my 'phone into my wired LAN. WAN access is via ADSL.

           So I am playing around with the various codecs as shown on my Grandstream 'phone.  Am I on the right track and if so does anyone know which codec I should be using?

If you're your using Asterisk then remember that Asterisk is free and therefore you don't get the decent codecs.

> Ehh, sorry my brain's abit overheated, should have been smoothwall > http://smoothwall.org/  ;) You could also set up a Asterisk box on a > spare PC, then you can illiminate all "outside" factors. > www.asterisk.org It's not so hard if you only want to experiment with > SIP, and gives you alot of other col functions, like voicemail, > conference etc. /Stefan

=Edit

> Do you have any QoS or bandwidth control running so that your voip traffic is prioritised. > You tend to find that Voip likes consistant, steady badnwidth and ping times. When you > say 128K do you mean at the broadband end in which case is it a contended service? > i.e. are you sure you're getting the uplink speed you think you are?

No QoS/bandwidth control in place, but of course "am I getting the speed I think I am" is valid since I use ADSL as backhaul.

> Using as decent codec like G.729 or G.723 your bandwidth demand should be around > 24K (bits per second). Thanks, I'll try G.723 with a frames/packet setting of 8. For info, I found a faq for the Grandstream BudgeTone 100 - http://www.grandstream.com/FAQ.htm. I can see that using the factory settings my bandwidth would have been chewed up since the default is for "toll" quality (64k).

btw - is anyone using an Asterisk/LW set-up?

I shall have a play - thanks all

Using as decent codec like G.729 or G.723 your bandwidth demand 

should be around > 24K (bits per second). If you're using some of the cra**ier 'free' codecs like G.711 then > this could be double or tripple. If your voip equipment has parameters to control the > frames/packet setting then try increasing this (upto 8 is usually okay). This will > significantly reduce your badnwidth demand. > > Do you have any QoS or bandwidth control running so that your voip traffic is prioritised. > You tend to find that Voip likes consistant, steady badnwidth and ping times. When you > say 128K do you mean at the broadband end in which case is it a contended service? > i.e. are you sure you're getting the uplink speed you think you are? > > On 26 Jul 2004 at 12:59, Mike Crompton wrote: >

> > I've been reading the recent questions regading VoIP with some > > interest. The following may have been covered previously, if so > > apologies. > > > > My testing is showing that the bandwidth required for my SIP phone is > > greater than I am expecting it to be. I am not looking for "studio" > > quality voice, in fact the complete opposite. A "tinny" sounding > > conversation would be good enough for me. What I do need is for the > > conversation to be uninterrupted. I figure that the lower quality the > > sound the less bandwidth required - is this assumption correct? > > > > The problem is that conversations I have are "broken up" unless I make > > around 128k available for the upstream link - this just doesn't agree > > with the stuff I have read where I am told that 32k should be enough, > > again bearing in mind with the quality I am prepared to live with. > > > > In my testing, I have ended up taking the wireless element out of the > > equation (why complicate things further!), instead just hooking my > > 'phone into my wired LAN. WAN access is via ADSL. > > > > So I am playing around with the various codecs as shown on my > > Grandstream 'phone. Am I on the right track and if so does anyone > > know which codec I should be using?

> > The problem is that conversations I have are "broken up" unless I > > make around 128k available for the upstream link - this just doesn't

> > agree with the stuff I have read where I am told that 32k should be > > enough, again bearing in mind with the quality I am prepared to live

> > with. > > > > In my testing, I have ended up taking the wireless element out of > > the equation (why complicate things further!), instead just hooking > > my 'phone into my wired LAN. WAN access is via ADSL. > > > > So I am playing around with the various codecs as shown on my > > Grandstream 'phone. Am I on the right track and if so does anyone > > know which codec I should be using?

At 3:56 pm -0700 6/8/04, Stephen Murphy wrote: >Thanks for your help. I set the box to 'Lock to Gateway' and bingo >everything works. However, the box (DN01) was set to 'Any' and it >was working perfectly. So far being behind a NAT does not seem to be >an issue as I have always had set "Gateway NAT workaround". Say I >have a repeater that is at the end of a 5 hop mesh and I set "Lock >to Gateway" where the Gateway is 5 hops away. Does the signal beam >directly to the Gateway or does it utilize the other repeaters and >create a mesh as it is designed to do.

It utilised the other nodes (as in creating a mesh), but the VOIP traffic goes over the tunnel, so it doesn't matter what other nodes are in between.

VOIP Setup on mesh Edit

Many of our access points (especially the high traffic ones) have been converted to dual radio boxes. We can use high-gain point-to-point links between nodes, and some nodes have a regular access point on the ethernet interface (wired captive portal turned on) and dhcp not used on the wlan interface(s). This setup helps out a lot with latency and throughput because you basically have full duplex at that box - making VoIP work well, even when the total distance covered from AP to gateway is 7 to 8 miles and 2 to 3 hops.

I know that it's all half-duplex at the interface, but locally at the AP, it's receiving on one interface and transmitting on the other - that is what I mean by full-duplex.


> Have clients trying commercial services on mesh with mixed results. > What is your layout like and how well does it work ?



LockTogateway Edit

{{{ Thanks for your help. I set the box to ‘Lock to Gateway’ and bingo everything works. However, the box (DN01) was set to ‘Any’ and it was working perfectly. So far being behind a NAT does not seem to be an issue as I have always had set “Gateway NAT workaround”. Say I have a repeater that is at the end of a 5 hop mesh and I set “Lock to Gateway” where the Gateway is 5 hops away. Does the signal beam directly to the Gateway or does it utilize the other repeaters and create a mesh as it is designed to do. Set "Lock to Gateway" to your gateway node on DN01,DN02. Also looks like your gateway is behind a NAT, try "Gateway NAT workaround" setting on GW. I didn't get VoIP to work until we put our GW on a public IP.

I have been having issues with my VOIP setup on one of my 2 Downlink Nodes. The VOIP phone connects and registers on my Gateway Node (GN) and my first Downlink Node (DN01), however I can’t register on my second Downlink Node (DN02). My setup is as follows:

Gateway Node (GN)

Box function: Always Gateway

Cross mesh routing: StandardIA

Gateway use DHCP dns: No

Daisy chain gateway's dns: Yes

static eth addr: 192.168.1.5

static eth netmask: 255.255.255.0

static eth gateway: 192.168.1.1

static eth dns: 192.168.1.1

Wired local 192.168.: 1.5

My Gateway node is a wired connection from my router and has been given a static address as outlined above.

DN01

Box function: Auto Detect

Cross mesh routing: Standard IA

Gateway use DHCP dns: No

Daisy chain gateway's dns: Yes

static eth addr:

static eth netmask:

static eth gateway:

static eth dns:

Wired local 192.168.: 2.5

Lock to gateway: Any

At the moment DN01 has no wired clients and works very well with the VOIP phone.

DN02

Box function: Auto Detect

Cross mesh routing: Standard IA

Gateway use DHCP dns: No

Daisy chain gateway's dns: Yes

static eth addr:

static eth netmask:

static eth gateway:

static eth dns:

Wired local 192.168.: 3.5

Lock to gateway: Any

The only difference between DN01 and DN02 is the wired local client field. As with DN01, DN02 does not have any wired clients. Can anyone see an obvious problem with the configuration of DN02. I have tried everything to connect to client DN02 with the VOIP phone without any luck. I have uninstalled and reinstalled the SIP proxy. All boxes are running on build 25dev87 and all have been working with the internet and email. }}}


asdf asd Edit

Ok here goes, this can get complicated if you think about it the wrong way. To start with configure node A to use the preferred gateway at node D, you will find this on the WiaNa settings. This will ensure that all traffic on node A uses Node D for its Internet traffic. Then set, in WiaNa again, the 2nd preferred gateway to node B. Next you need to set the following, again in wiana Disabled failed upstream, which is in the extra features menu

Then in Custom local test host: you need to put an ip address in one that is part of your isp but before the internet. For example do a Tracert in cmd promt for www.bbc.co.uk then you will see a list of ipaddresses. I use tiscali, so …. I picked an ip address that was still part of the tiscali network, there for it must be a RouTer or a firewall, or something else belonging to tiscali. Input this number into the field.

Then if your ISP fails or your connection to it fails the MeshAP will not be able to ping it and turn itself into a RepeaTer. So in your scenario node A is using node D for its internet traffic. Node D is checking that it can see your ISP router, the ISP router goes down or there is a lockup on your local router. Node D can not see the ip address any more so it changes to a RepeaterNode.

Node A then sees that node D is no longer a GateWayNotes and decides to go to its 2nd preferred gateway and once again internet is re-established. Be ware though – the node D will not revert back into a gateway unless it is rebooted and can pick up an ip address!


=Edit

I have 5 nodes, A, B, C, D and E. I have 2 backhaul routes. One is ADSL the other is SateLlite. Node B is one hub with ADSL connected and D is the other with Satellite. I want all users traffic on A to go through the satellite which is on D. A can only see D "through" B. Will the traffic be routed correctly or will it take the shortest route i.e. using B's ADSL backhaul? Of course if the satellite should go down, I want the traffic to go through B's ADSL backhaul route.


VoipAndWrt54g Edit

Also would be worth checking that the Handytone 486 actually works with the exchange you are connecting it to, I recently wasted a couple of hours trying to get it to dial an Asterisk@home 1.5 server to find it would not work until I used asterisk@home 2.5. Try a softphone as an alternative 1st, I've have good results with SJPhone on my Mac. This way you can rule the Mesh in or out of the equation.



I am trying to get a Grandstream handytone 486 to work thru a gateway node on my test network. I have gone thru the setup on the wiki and also looked thru old mailing list posts but no luck. I am using a wrt54g to connect to my mesh gateway. The wrt is set to get ip dhcp from the gateway and I am putting out dhcp to the grandstream. The wrt setup works on another non mesh ap I have here that goes to the same provider, so I must be missing something in the mesh gateway setup. Any help would be greatly appreciated. I am usig dev106 on this gateway.


MoskalukNotes Edit

We have been operating Asterisk, Asterisk@home (now trixbox) and Bicoms commercial offering PBXware for two years now. ADR Communications Ltd are deployment engineers for Telappliant.com or voiptalk.co.uk working both closely with bicomsystems and digium.

If anyone should find there self's in trouble setting up asterisk, then please feel free to drop me a line. Adrian Robinson


I have written 3 new articles about using Asterisk Telephony application in Wireless Mesh. One is an Introduction of Asterisk (Trixbox) and Locustworld call Locusts meet the Hare. The second one is similar to the original wireless Topology page call VoIP using Wireless Mesh Infrastruture. And finally a little DIY article of how to install a radio card in a TrixBox call Wireless Setup for a TrixBox.

For people who have been building their own infrastructure the next thing you should explore is setting up your own VoIP applications with in the wireless mesh. Maybe you have in mind replacing an existing Office PBX, or having several extensions at home and handling calls more slickly or starting up your own Internet Telephone Company. Either way there is one piece of software which coupled with the right hardware provides a comprehensive PBX solution, and that is Asterisk.

   For people who have built their own telephone system using Asterisk you can create or extend an application like VoIP through Wi-Fi. Specifically using Wireless Mesh Locustworld and Trixbox (Asterisk) can create some exciting new opportunities. The Do-It-Yourself (DIY) articles and references are located on Research Papers. 

http://www.moskaluk.com/voip_using_wireless_mesh_infrast.htm

http://www.moskaluk.com/locusts_meet_the_hare.htm

http://www.moskaluk.com/wireless_network_setup_for_trixbox.htm


Links Edit

Part of MeshNetworking

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