I'm getting a little frustrated with RTP relay which does not seem to work when the supposed requirements, two devices with the same codec, are talking to each other. I do not know how to go about troubleshooting this. Does anyone know of why this might not be working as it should??
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VCMs used when RTP Relay should kick in.
- Thread starter clindberg
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- #2
I have an H323 trunk connecting to a third party H323 gateway. When the phones make an outbound connection, they connect just fine, but the call detail screen says that it is a VCM connection rather than RTP relay, and 2 VCM channels show in use. Both codecs are g711u....
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- #3
kwing112000
Vendor
how are your check boxes for direct media path. they should be checked off to make this work. can the phone ping out to the H323 gateway?
Kevin Wing
ACSS Small and Medium Enterprise (SME) Communications
ACS- Implement IP Office
ACA- Implement IP Office
Carousel Industries
Kevin Wing
ACSS Small and Medium Enterprise (SME) Communications
ACS- Implement IP Office
ACA- Implement IP Office
Carousel Industries
intrigrant
Systems Engineer
I was told long ago by Avaya developers that H.323 trunks always use VCM resources, i don't think that has ever changed. This does not count for SIP trunks though.
amriddle01
Programmer
The following from the H323 line settings Help file would suggest otherwise:
Allow Direct Media Path: Default = On
This settings controls whether H323 calls must be routed via the H323 gatekeeper (the IP Office) or can be routed alternately if possible within the network structure.
· If enabled, H323 calls can take routes other than through the IP Office. This removes the need for a voice compression channel. Both ends of the calls must support Direct Media. Enabling this option may cause some vendors problems with changing the media path in mid call. Pre-IP Office 4.0, when using direct media path, it is not always possible for the extension to be recorded or monitored.
· If disabled or not supported at on one end of the call, the call is routed via the IP Office.
· On pre-4.0 these calls would then require a voice compression channel even if the IP devices use the same audio codec.
· On IP Office 4.0 and higher, RTP relay support allows calls between devices using the same audio codec to not require a voice compression channel.
And as I say that's specifically referring to H323 trunks so it looks like H323 relay should work
ACSS (SME)
APSS (SME)
Allow Direct Media Path: Default = On
This settings controls whether H323 calls must be routed via the H323 gatekeeper (the IP Office) or can be routed alternately if possible within the network structure.
· If enabled, H323 calls can take routes other than through the IP Office. This removes the need for a voice compression channel. Both ends of the calls must support Direct Media. Enabling this option may cause some vendors problems with changing the media path in mid call. Pre-IP Office 4.0, when using direct media path, it is not always possible for the extension to be recorded or monitored.
· If disabled or not supported at on one end of the call, the call is routed via the IP Office.
· On pre-4.0 these calls would then require a voice compression channel even if the IP devices use the same audio codec.
· On IP Office 4.0 and higher, RTP relay support allows calls between devices using the same audio codec to not require a voice compression channel.
And as I say that's specifically referring to H323 trunks so it looks like H323 relay should work
ACSS (SME)
APSS (SME)
amriddle01
Programmer
intrigrant
Systems Engineer
Yes, maybe it is changed, i was dealing with this in 1.4 software and since then i just make sure i have enough resources and never looked at it again.
I haven't verified this recently (ie in R6) but I did notice that the VCM usage is inconsistent. Sometimes it would use RTP Relay and other times the VCM between SIP and Avaya 5610s. Same phone and same trunk and same codec but different results on individual calls.
I just wrote it off as typical Avaya not knowing what they want to do.
Also, it was my understanding that RTP relay is not 100% the same as direct media? Is my below assumption correct?
VCM = VCM Usage, media is going through the VCM hardware
RTP Relay = media packets are still routed through the IP Office but doesn't process/convert with the VCM
Direct Media = media packets are device to device
I just wrote it off as typical Avaya not knowing what they want to do.
Also, it was my understanding that RTP relay is not 100% the same as direct media? Is my below assumption correct?
VCM = VCM Usage, media is going through the VCM hardware
RTP Relay = media packets are still routed through the IP Office but doesn't process/convert with the VCM
Direct Media = media packets are device to device
- Thread starter
- #10
amriddle01
Programmer
nlasch, you are correct with the VCM/system usage assumptions, don't know why it's inconsistant though...do Avaya?
ACSS (SME)
APSS (SME)
ACSS (SME)
APSS (SME)
I have noticed differences on SIP trunks and VCM usage before. Some SIP providers use loads of VCMs for just for calls, ome need to use one to put calls on hold, some don't use as many VCMs when CODEC's match but others do.
Pain in the ass!! I did ask 3 years ago for an Avaya blueprint of how many VCMs should be used at each point. Still waiting for Avaya to decide for themselves I think!!!
Clindberg, I assume you have IP Phones??? Are the phones hard coded to use the same CODEC as the H323 gateway and the H323 line. Maybe you could use Wireshark to check this is actually hapening??
Jamie Green
Football is not a matter of life and death-It is far more important!!!!
Pain in the ass!! I did ask 3 years ago for an Avaya blueprint of how many VCMs should be used at each point. Still waiting for Avaya to decide for themselves I think!!!
Clindberg, I assume you have IP Phones??? Are the phones hard coded to use the same CODEC as the H323 gateway and the H323 line. Maybe you could use Wireshark to check this is actually hapening??
Jamie Green
Football is not a matter of life and death-It is far more important!!!!
latitude12
Technical User
It makes me think that there is other criteria out there other than simply "same codec". Are there different versions of g.711, or did I just ask a stupid question."
Yep, G711 a-law and mu-law, also packet size can vary. Difference in either of these settings would require transcoding (i.e. a VCM would be used). This may also be required if silence suppression is set differently at either end, but I'm not sure about that.
Yep, G711 a-law and mu-law, also packet size can vary. Difference in either of these settings would require transcoding (i.e. a VCM would be used). This may also be required if silence suppression is set differently at either end, but I'm not sure about that.
latitude12
Technical User
IPO uses 20ms packet size for G771 (I think). Also, like Jamie asked, what phone(s) are you using?
- Thread starter
- #15
From the knowledgebase:
When are Voice Compression Channels Used
The voice compression channels are used as follows. • IP Device to Non-IP Device
These calls require a voice compression channel for the duration of the call. If no channel is available, busy indication is returned to the caller.
• IP Device to IP Device
• Call progress tones (for example dial tone, secondary dial tone, etc) do not require voice compression channels with the following exceptions:
• Short code confirmation, ARS camp on and account code entry tones require a voice compression channel.
• Devices using G723 require a voice compression channel for all tones except call waiting.
• When a call is connected:
• If the IP devices use the same audio codec no voice compression channel is used.
• If the devices use differing audio codecs, a voice compression channel is required for each.
• Non-IP Device to Non-IP Device
No voice compression channels are required except for Small Office Edition Embedded Voicemail access.
• Music on Hold
This is provided from the IP Office's TDM bus and therefore requires a voice compression channel when played to an IP device.
• Conference Resources and IP Devices
Conferencing resources are managed by the conference chip which is on the IP Office's TDM bus. Therefore, a voice compression channel is required for each IP device involved in a conference. This includes services that use conference resources such as call listen, intrusion, call recording and silent monitoring.
• Page Calls to IP Device
Page calls require 1 voice compression channel per audio codec being used by any IP devices involved. IP Office 4.0 and higher only uses G729a for page calls, therefore only requiring one channel but also only supporting pages to G729a capable devices.
• Voicemail Services and IP Devices
Calls to the IP Office voicemail servers are treated as data calls from the TDM bus. Therefore calls from an IP device to voicemail require a voice compression channel.
• On the Small Office Edition, embedded voicemail uses voice compression channels for audio conversion. Therefore all calls to Small Office Edition embedded voicemail require a voice compression channel and calls from IP devices require two voice compression channels.
• Fax Calls
These are voice calls but with a slightly wider frequency range than spoken voice calls. IP Office only supports fax across IP between IP Office systems with the Fax Transport option selected. It does not currently support T38.
• SIP Calls
• SIP Line Call to/from Non-IP Devices
Voice compression channel required.
• Outgoing SIP Line Call from IP Device
No voice compression channel required.
• Incoming SIP Line Call to IP Device
Voice compression channel reserved until call connected.
• T38 Fax Calls
IP Office Release 5+ supports T38 fax on SIP trunks and SIP extensions. Each T38 fax call uses a VCM channel.
• Within a Small Community Network, an T38 fax call can be converted to a call across across an H323 SCN lines using the IP Office Fax Transport Support protocol. This conversion uses 2 VCM channels.
• In order use T38 Fax connection, the Equipment Classification of an analog extension connected to a fax machine can be set Fax Machine. Additionally, a new short code feature Dial Fax is available.
Note: T3 IP devices must be configured to 20ms packet size for the above conditions to apply. If left configured for 10ms packet size, a voice compression channel is needed for all tones and for non-direct media calls.
Measuring Channel Usage
The IP Office System Status Application can be used to display voice compression channel usage. Within the Resources section it displays the number of channel in use. It also displays how often there have been insufficient channels available and the last time such an event occurred.
Homo sapiens non urinat in ventum
honey, i fried the IP Office !!!
Sarcasm, it's only one of the services I offer.
When are Voice Compression Channels Used
The voice compression channels are used as follows. • IP Device to Non-IP Device
These calls require a voice compression channel for the duration of the call. If no channel is available, busy indication is returned to the caller.
• IP Device to IP Device
• Call progress tones (for example dial tone, secondary dial tone, etc) do not require voice compression channels with the following exceptions:
• Short code confirmation, ARS camp on and account code entry tones require a voice compression channel.
• Devices using G723 require a voice compression channel for all tones except call waiting.
• When a call is connected:
• If the IP devices use the same audio codec no voice compression channel is used.
• If the devices use differing audio codecs, a voice compression channel is required for each.
• Non-IP Device to Non-IP Device
No voice compression channels are required except for Small Office Edition Embedded Voicemail access.
• Music on Hold
This is provided from the IP Office's TDM bus and therefore requires a voice compression channel when played to an IP device.
• Conference Resources and IP Devices
Conferencing resources are managed by the conference chip which is on the IP Office's TDM bus. Therefore, a voice compression channel is required for each IP device involved in a conference. This includes services that use conference resources such as call listen, intrusion, call recording and silent monitoring.
• Page Calls to IP Device
Page calls require 1 voice compression channel per audio codec being used by any IP devices involved. IP Office 4.0 and higher only uses G729a for page calls, therefore only requiring one channel but also only supporting pages to G729a capable devices.
• Voicemail Services and IP Devices
Calls to the IP Office voicemail servers are treated as data calls from the TDM bus. Therefore calls from an IP device to voicemail require a voice compression channel.
• On the Small Office Edition, embedded voicemail uses voice compression channels for audio conversion. Therefore all calls to Small Office Edition embedded voicemail require a voice compression channel and calls from IP devices require two voice compression channels.
• Fax Calls
These are voice calls but with a slightly wider frequency range than spoken voice calls. IP Office only supports fax across IP between IP Office systems with the Fax Transport option selected. It does not currently support T38.
• SIP Calls
• SIP Line Call to/from Non-IP Devices
Voice compression channel required.
• Outgoing SIP Line Call from IP Device
No voice compression channel required.
• Incoming SIP Line Call to IP Device
Voice compression channel reserved until call connected.
• T38 Fax Calls
IP Office Release 5+ supports T38 fax on SIP trunks and SIP extensions. Each T38 fax call uses a VCM channel.
• Within a Small Community Network, an T38 fax call can be converted to a call across across an H323 SCN lines using the IP Office Fax Transport Support protocol. This conversion uses 2 VCM channels.
• In order use T38 Fax connection, the Equipment Classification of an analog extension connected to a fax machine can be set Fax Machine. Additionally, a new short code feature Dial Fax is available.
Note: T3 IP devices must be configured to 20ms packet size for the above conditions to apply. If left configured for 10ms packet size, a voice compression channel is needed for all tones and for non-direct media calls.
Measuring Channel Usage
The IP Office System Status Application can be used to display voice compression channel usage. Within the Resources section it displays the number of channel in use. It also displays how often there have been insufficient channels available and the last time such an event occurred.
Homo sapiens non urinat in ventum
honey, i fried the IP Office !!!
Sarcasm, it's only one of the services I offer.
amriddle01
Programmer
Thanks for that tlpeter, but it just confirms that it should be doing RTP relay but isn't
ACSS (SME)
APSS (SME)
ACSS (SME)
APSS (SME)
- Thread starter
- #19
Although something in a couple of your posts begs further research,
"Note: T3 IP devices must be configured to 20ms packet size for the above conditions to apply. If left configured for 10ms packet size, a voice compression channel is needed for all tones and for non-direct media calls. "
I'm wanting to see if I can tweak my gateway packet size.
20ms packet size refers to how much "voice time" is encoded per packet I assume.
"Note: T3 IP devices must be configured to 20ms packet size for the above conditions to apply. If left configured for 10ms packet size, a voice compression channel is needed for all tones and for non-direct media calls. "
I'm wanting to see if I can tweak my gateway packet size.
20ms packet size refers to how much "voice time" is encoded per packet I assume.
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