FreePBX Direct RTP Setup
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I will start this off by saying that I haven't the slightest idea what I'm doing. I'm learning as I go.
So, the other day I stumbled across configuring FreePBX to hand-off the RTP stream to the end devices so that audio is direct between peers (when possible, of course). I have a test server set up with a handful of phones registered to it. Initiating calls between extensions works beautifully and direct audio works as expected.
However, I'm running into a snag with transferring calls. For some reason, call transfers do not work properly and result in one way audio. Here is an example situation:
1 calls 2. Call audio is perfectly fine. 2 transfers the call to 3. 3 can hear 1, but 1 cannot hear 3.
For what it's worth, the phone make/model does not seem to change the result. I've used Polycom SoundPoint 321's, 331's, Yealink T25G's, and Cisco SPA 525G's in my testing and all have the same result when transferring a call.
I have things set up as detailed here: https://sashikasuren.blogspot.pt/2013/08/peer-to-peer-communication-in-freepbx.html
Any ideas?
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Is the call a blind transfer or did 2 talk to 3 first? Did that portion of the process have 2 way audio?
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@JaredBusch I've only done blind transfers. During the transfer 1 heard our hold music if that makes a difference. I'll try with 2 and 3 talking before making the transfer and report back.
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@JaredBusch said in FreePBX Direct RTP Setup:
Is the call a blind transfer or did 2 talk to 3 first? Did that portion of the process have 2 way audio?
Ok, I did the following:
1 called 2 - two way audio
2 pressed the transfer soft key and called 3. 3 answered and there was two way audio between 2 and 3.
2 transferred the cal to 3. 3 could hear 1, but 1 could not hear 3.
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@anthonyh said in FreePBX Direct RTP Setup:
@JaredBusch I've only done blind transfers. During the transfer 1 heard our hold music if that makes a difference. I'll try with 2 and 3 talking before making the transfer and report back.
I am not sure if that is a good sign or not without seeing full logs and having something setup to test myself, but it should be. That music comes from the PBX and if the audio stream was still only phone to phone, you would not hear it.
I've never bothered to enable reinvite anywhere before so I am not clear on the exact flow of the audio stream when a reinvited call gets put on hold.
Why are you bothering with this by the way? Is there some bandwidth constraint causing the need for this functionality?
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@JaredBusch said in FreePBX Direct RTP Setup:
@anthonyh said in FreePBX Direct RTP Setup:
@JaredBusch I've only done blind transfers. During the transfer 1 heard our hold music if that makes a difference. I'll try with 2 and 3 talking before making the transfer and report back.
I am not sure if that is a good sign or not without seeing full logs and having something setup to test myself, but it should be. That music comes from the PBX and if the audio stream was still only phone to phone, you would not hear it.
I've never bothered to enable reinvite anywhere before so I am not clear on the exact flow of the audio stream when a reinvited call gets put on hold.
Why are you bothering with this by the way? Is there some bandwidth constraint causing the need for this functionality?
From what I understand the MOH is fine. Asterisk takes over the audo stream when necessary (to inject MOH for instance) from what I've read. I can confirm that it is working, as I can initiate a call between extensions that are registered on phones on the same physical network switch and same subnet, then unplug the uplink on said switch and proceed with the call like nothing happens. In that scenario placing the call on hold yields silence.
Again, I'm learning as I go...
No, there is no specific necessity for the feature, but if I can make it work properly I'd like to use it. Every little bit helps, and if I can keep the audio of intra-office calls from traversing WAN links, I'd like to do so.
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@anthonyh said in FreePBX Direct RTP Setup:
@JaredBusch said in FreePBX Direct RTP Setup:
@anthonyh said in FreePBX Direct RTP Setup:
@JaredBusch I've only done blind transfers. During the transfer 1 heard our hold music if that makes a difference. I'll try with 2 and 3 talking before making the transfer and report back.
I am not sure if that is a good sign or not without seeing full logs and having something setup to test myself, but it should be. That music comes from the PBX and if the audio stream was still only phone to phone, you would not hear it.
I've never bothered to enable reinvite anywhere before so I am not clear on the exact flow of the audio stream when a reinvited call gets put on hold.
Why are you bothering with this by the way? Is there some bandwidth constraint causing the need for this functionality?
From what I understand the MOH is fine. Asterisk takes over the audo stream when necessary (to inject MOH for instance) from what I've read. I can confirm that it is working, as I can initiate a call between extensions that are registered on phones on the same physical network switch and same subnet, then unplug the uplink on said switch and proceed with the call like nothing happens. In that scenario placing the call on hold yields silence.
Again, I'm learning as I go...
No, there is no specific necessity for the feature, but if I can make it work properly I'd like to use it. Every little bit helps, and if I can keep the audio of intra-office calls from traversing WAN links, I'd like to do so.
You are spending a lot of time to save 200kbps per intra office call. How much is this project costing versus how much will it benefit your company?
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@JaredBusch said in FreePBX Direct RTP Setup:
You are spending a lot of time to save 200kbps max per intra office call. How much is this project costing versus how much will it benefit your company?
FTFY. Might be far less and it's split between the directions. And you give up some features like call control, monitoring, recording and reporting. Sure those might be trivial, but I'd think hard about @JaredBusch's comment here. Typically intraoffice calling uses an unnoticeably small bandwidth. It's totally trivial.
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@scottalanmiller said in FreePBX Direct RTP Setup:
@JaredBusch said in FreePBX Direct RTP Setup:
You are spending a lot of time to save 200kbps max per intra office call. How much is this project costing versus how much will it benefit your company?
FTFY. Might be far less and it's split between the directions. And you give up some features like call control, monitoring, recording and reporting. Sure those might be trivial, but I'd think hard about @JaredBusch's comment here. Typically intraoffice calling uses an unnoticeably small bandwidth. It's totally trivial.
These are good points. Our FreePBX instance is regularly sending and receiving 3-4 Mbps of traffic, which isn't really that much. I'll have to see how much of this is internal/internal vs internal/external to see what sort of difference this would make.
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@anthonyh said in FreePBX Direct RTP Setup:
@scottalanmiller said in FreePBX Direct RTP Setup:
@JaredBusch said in FreePBX Direct RTP Setup:
You are spending a lot of time to save 200kbps max per intra office call. How much is this project costing versus how much will it benefit your company?
FTFY. Might be far less and it's split between the directions. And you give up some features like call control, monitoring, recording and reporting. Sure those might be trivial, but I'd think hard about @JaredBusch's comment here. Typically intraoffice calling uses an unnoticeably small bandwidth. It's totally trivial.
These are good points. Our FreePBX instance is regularly sending and receiving 3-4 Mbps of traffic, which isn't really that much. I'll have to see how much of this is internal/internal vs internal/external to see what sort of difference this would make.
That is easy. Just open the Dashboard and look at the trunk usage. That tells you right there how many calls are going in out out the external pipe. You have ~10 calls for each 1mbps.
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@scottalanmiller said in FreePBX Direct RTP Setup:
@JaredBusch said in FreePBX Direct RTP Setup:
You are spending a lot of time to save 200kbps max per intra office call. How much is this project costing versus how much will it benefit your company?
FTFY. Might be far less and it's split between the directions. And you give up some features like call control, monitoring, recording and reporting. Sure those might be trivial, but I'd think hard about @JaredBusch's comment here. Typically intraoffice calling uses an unnoticeably small bandwidth. It's totally trivial.
Now, while the CALL may be 80kbps, once you tack in overhead, there overall usage is easily 100kbps. I can look at any number of systems deployed in the wild and watch the traffic go up and down by almost exactly 100kbps.
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Here's a screen shot of the dashboard. How do I interpret what I'm seeing?
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Why the heck does it show the network in KB instead of Kb. How silly.
That shows your average at pretty close to 100Kb/s. Just eyeballing it.
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Total server traffic is 3.7 Mb/s. 36 calls at the moment.
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@scottalanmiller said in FreePBX Direct RTP Setup:
Why the heck does it show the network in KB instead of Kb. How silly.
That shows your average at pretty close to 100Kb/s. Just eyeballing it.
No idea. It's been this way since I've been here.
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@anthonyh said in FreePBX Direct RTP Setup:
@scottalanmiller said in FreePBX Direct RTP Setup:
Why the heck does it show the network in KB instead of Kb. How silly.
That shows your average at pretty close to 100Kb/s. Just eyeballing it.
No idea. It's been this way since I've been here.
WTF is up with your calls counts not even being close? Do you do a lot of internal conference calling?
You have 36 active calls but only 6 internal and 5 external? Where in the hell are the other 25?
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@anthonyh said in FreePBX Direct RTP Setup:
@scottalanmiller said in FreePBX Direct RTP Setup:
Why the heck does it show the network in KB instead of Kb. How silly.
That shows your average at pretty close to 100Kb/s. Just eyeballing it.
No idea. It's been this way since I've been here.
FreePBX has always used that notation.
