This article describes how issues related to the RTP Packetization interval can be solved in the case of conference calls on VoipNow servers. The recommended solution is to make sure that all parties involved in that call have the packetization interval set to 20ms during conference calls on VoipNow servers. Otherwise, when channels are bridged, voice breaks, missing or dropped syllables, unexplained clicks or noises may occur.
Overview
Voice breaks, missing or dropped syllables, unexplained clicks or noises are triggered by different values of the ptime parameter when channels are bridged. According to the RFC3551 for packetized audio, the default packetization interval must be 20 ms or one frame, whichever is longer, unless required otherwise. However, a receiver should be able to accept packages with a packetization interval between 0 and 200ms.
Good to know
If autoframing=yes is set in sip.conf within the global section, all calls will try to set the packetization based on the preferences of the remote end point.
When autoframing is enabled at global level, it applies to all regardless of the selected codec packetization.
If one of the parties involved in a call does not provide a ptime interval, the recommended value that should be used is 20ms.
In VoipNow, autoframing is not enabled in sip.conf. Consequently, VoipNow will use default values for ptime, depending on the selected codec.
For example, if a call is made using G711 as codec, the ptime will be 20ms. This means that each RTP packet sent during that call will only have 20ms of audio recorded in it.
Ptime and different devices
The ptime parameter for codecs can be set through provisioning by adding a specific value for each codec in the Device Configuration File #3:
voice.audioProfile.X.payloadSize="20"
The value for the payload size is in milliseconds: 10, 20, 30,...80;
X can be replaced by G711Mu, G711A, G722, G7221, and G729AB.
For example, when setting a ptime=20 for the G711A codec, the following line should be added to the provisioning template:
voice.audioProfile.G711A.payloadSize="20"
Aastra 480i
- To set the
ptimeparameter on a Aastra 480i codec, you need to go to the Advanced Settings->Global SIP->RTP Settings area; - Enter a Customized Codec Preference List which allows you to use the codecs that you prefer for this IP phone.
For example:
payload=8;ptime=20; payload=0;ptime=20;
Please note that payload 0 is for G.711 u-Law and payload 8 is for G.711 a-Law.
Aastra 6751i
- To set the RTP packetization interval on Aastra 6751i, you need to go to the Advanced Settings->Global SIP->Codec Preference List;
- Valid values: 5 to 90, in increments of 5 milliseconds;
- To use this phone registered on a VoipNow server, you need to set the Packetization interval to 20ms;
Grandstream Gxp2020
Web Interface
Go to Advanced Settings -> iLBC frame size and select a framesize of 30ms.
Provisioning Template
You need to set the P97 variable to 1 as shown below:
# iLBC Frame Size. 0 - 20ms(default), 1 - 30ms. P97 = 1
Yealink T20P
The ptime parameter can be set via provisioning templates.
In the device configuration file 1, you can set the ptime value as shown below:
[ account ] path=/config/voip/sipAccount0.cfg <yealink t20p configuration parameters> ptime = 20 [ account ] path=/config/voip/sipAccount1.cfg <yealink t20p configuration parameters> ptime = 20
Linksys PAP2
Some Linksys devices, such as PAP2 or SPA2102, have the RTP Packet Size parameter set to 30ms by default.
In such cases, the RTP Packet Size parameter can be changed from the SIP tab of the web interface.
For most users, the 0.030 factory default preset should be replaced with 0.020.
Please note that the RTP Packet Size parameter applies to all the lines served through that adapter.
Siemens Speedstream 3610
To set the RTP packetization interval on Siemens Speedstream 3610, you need to go to the Voice Setup -> Advanced Setting.
If you are using this device registered on a VoipNow server, please set the RTP Packetization period to 20ms. Valid values: in increments of 10ms.
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