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[bjbrashier at gmail.com]

I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy).
 
I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call.
 
My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace?

[msc at freeswitch.org]

Can you describe your networking environment a bit? One thing that can affect the latency of your voice traffic is your network infrastructure. If you can isolate FS and some phones on a separate, controlled network then possibly you can start narrowing it down to other factors.

-MC

On Tue, Jun 16, 2009 at 10:51 AM, Bradley Brashier <bjbrashier@gmail.com (bjbrashier@gmail.com)> wrote:

[anthony.minessale at g...]

The problem comes from the timing of certain phones during the capture of audio actually clocked slightly faster than what it advertises.
Try the latest trunk with all the defaults in your sip profile as we have tried to make the defaults deal with this automatically.


On Tue, Jun 16, 2009 at 12:51 PM, Bradley Brashier <bjbrashier@gmail.com (bjbrashier@gmail.com)> wrote:

[bjbrashier at gmail.com]

I have two different network setups, and have seen similar lag on both.
 
The first is my home testbed. I'm connected to the internet through a home router and then a cablemodem. The home environment is pretty spare, of course. 2 machines and a couple of T-mobile cell phones with their SIP communication is all that goes through there. I have used my cell phone, a couple of different softphones, Gizmo call-ins, and regular PSTN calls. The worst lag is the T-mobile cell phones, but I'm happy to write that off as T-mobile's problem if we'd like.
 
The second is the debug server environment on the systems where the conference product will eventually reside. The system is very complex, as it is already running a major hosted PBX service written years ago. I'm afraid all of the details of this system are beyond me, but I know that it includes a PSTN gateway, more T1s than I can count, and I'm having to split the RTP and SIP packets on separate ports for security and organizational purposes. For call-ins, I have used T-mobile again and regular PSTN, no softphones (yet). Obviously, this is the important environment, and the PSTN lag is somewhere around 500-700 ms (subjective).
 
So am I correct in understanding that this is not a common issue, then, and that something can theoretically be done to help it?


On Tue, Jun 16, 2009 at 11:35 AM, Michael Collins <msc@freeswitch.org (msc@freeswitch.org)> wrote:

[joshm at wabashcenter.com]

I am not as knowledgeable as the developers that will respond to your question but I had the same problem as you. Here is what I did to combat the delay:

First off I started everything from scratch. I reinstalled Linux and then I reinstalled FreeSWITCH by creating .deb packages.
I then created my own conference profile and set the sample rate to 4000 and changed the energy level to 20.
I also made sure to test the conference room from phones that were in completely different areas so there wasn’t a chance for feedback or really bad echoing problems.

Once I knew the delay was solved I raised the sample rate to 8000. I tested it to make sure it would work properly.

As Michael stated, this could be your network infrastructure but I just wanted to let another FreeSWITCH user know what I did to try and stop the voice delay.

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Bradley Brashier
Sent: Tuesday, June 16, 2009 1:52 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Voice lag in conference


I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy).

I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call.

My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace?
This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version.

[bjbrashier at gmail.com]

I'm not sure I've got the opportunity to do that at the moment, but I do appreciate the point of view of a fellow product user. Were you able to eliminate noticeable lag, or just reduce it to reasonable levels?
 
I'll try to do something similar when I update to the newest trunk as Anthony suggested. My copy is only a week old, but I'll try whatever has a chance of working, and I know you guys have been working on conferencing (the Moderator function couldn't have been timed better for me!).


On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon <joshm@wabashcenter.com (joshm@wabashcenter.com)> wrote:

[joshm at wabashcenter.com]

I was able to reduce it considerably. I can’t say it is completely gone but I am very confident the ~.5 second delay I hear is because of the time it takes my voice to go through the leaps and bounds of the phone company to our server. I had at least a 3-5 second delay before I experimented with the conference settings.

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Bradley Brashier
Sent: Tuesday, June 16, 2009 5:02 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Voice lag in conference


I'm not sure I've got the opportunity to do that at the moment, but I do appreciate the point of view of a fellow product user. Were you able to eliminate noticeable lag, or just reduce it to reasonable levels?



I'll try to do something similar when I update to the newest trunk as Anthony suggested. My copy is only a week old, but I'll try whatever has a chance of working, and I know you guys have been working on conferencing (the Moderator function couldn't have been timed better for me!).

On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon <joshm@wabashcenter.com (joshm@wabashcenter.com)> wrote:
I am not as knowledgeable as the developers that will respond to your question but I had the same problem as you. Here is what I did to combat the delay:

First off I started everything from scratch. I reinstalled Linux and then I reinstalled FreeSWITCH by creating .deb packages.
I then created my own conference profile and set the sample rate to 4000 and changed the energy level to 20.
I also made sure to test the conference room from phones that were in completely different areas so there wasn’t a chance for feedback or really bad echoing problems.

Once I knew the delay was solved I raised the sample rate to 8000. I tested it to make sure it would work properly.

As Michael stated, this could be your network infrastructure but I just wanted to let another FreeSWITCH user know what I did to try and stop the voice delay.

From: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] On Behalf Of Bradley Brashier
Sent: Tuesday, June 16, 2009 1:52 PM
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Subject: [Freeswitch-users] Voice lag in conference


I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy).

I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call.

My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace?


This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version.


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This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version.

[anthony.minessale at g...]

don't forget to read my suggestion too from earlier today =D


On Tue, Jun 16, 2009 at 4:30 PM, Josh Moon <joshm@wabashcenter.com (joshm@wabashcenter.com)> wrote:

[bjbrashier at gmail.com]

Will do, just haven't had the time, yet!

On Tue, Jun 16, 2009 at 2:55 PM, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:

[bjbrashier at gmail.com]

So I rebooted, installed some OS updates, synched up, and am running again. I've also been doing closer comparisons between the conference I'm running and the same phones through VOIP to other locations (like between the phones without the conference).
 
The lag isn't as bad as it was, a significant portion is due to the VOIP connection we've got (ie. conference aside), and yet certain phones still have more trouble through the conference than not, to the tune of at least 400ms more than the others.
 
At this point, I'm prepared to punt -- blame the specific phones for now, and look at it again in a month or so when the project is closer to "done".  But if anyone has any ideas on why certain phones would behave worse than others (a Polycom SoundPoint IP 320 SIP phone is the worst) I'm all ears.

BB

On Tue, Jun 16, 2009 at 2:58 PM, Bradley Brashier <bjbrashier@gmail.com (bjbrashier@gmail.com)> wrote: