How do I configure media channels for VoIP?

Export to PDF | Export to DOC

Cause:
Once a call has been accepted by an H.323 endpoint, two voice channels (media channels) are opened, one for each end of the call to transmit voice over.

For the call to be completely successful both channels must be of the same compression or companding type. In most cases the VoIP gateway will wait for an incoming channel to be established by the other party, and then duplicate an outgoing channel of the same type. This process is called channel mimicing. If for some reason the incoming channel open is delayed then an attempt is made to open an outgoing channel of the type specified in the h245.preferred_index field.

Although the gateway can support both G.723.1 and G.729A compression standards, only one can be selected at a time. The VoIP Gateway always loads the compression standard specified by the first entry in the h245.cap.n.name list.

G.711 companding is always available regardless of compression CODEC used.

Solution:
Sometimes it may be useful to change some of the DSP (digital signal processor for voice) parameters in order to improve the quality of the sound heard.

The table below gives an indication of what effect changing the DSP parameters has on the audible sound quality. See the DSP protocol parameters section for more information, and details on the allowable ranges for the variables below.

 

VoIP Gateway
DSP Parameter

Description

Effect of increasing this parameter

OpenPhone Audio Codec Setting

packet_time

size of voice packets transmitted by the VoIP Gateway
(milliseconds)

1) improves reception on busy reliable networks by decreasing the number of packets transmitted per second2) increases the likelihood of audible sound loss on unreliable networks

Signal

VP_FIFO_nom_delay

minimum jitter buffer size
(milliseconds)

improves audibility when interworking with software based codecs (e.g. Microsoft Netmeeting) which introduce permanent jitter. This also increases the delay for the voice path accordingly

Minimum Jitter

VP_FIFO_max_delay

maximum jitter buffer size
(milliseconds)

improves the audibility on unreliable data networks which introduce random amounts of jitter. In extreme cases this will increase voice path delay

Maximum Jitter

echo_tail_size

amount of echo cancellation used
(milliseconds)

eliminates echo, but introduces fixed length delays

N/A

VADU_enable_flag

silence suppression enable/disable

enabling will introduce a slight voice path delay, but result in packet suppression on the network when noone is speaking.

Silence
Checked/Un-Checked

VADU_threshold

silence suppression activation threshold (dBm)

increases the level at which the codec will differentiate between background noise and speech.

N/A

idle_noise_level

background “comfort” noise level

increases the level of ambient noise generated in the listeners ear when silence is detected at the sending end

N/A

tx_gain

packet transmit gain (dBm)

increases the sound level for packets transmitted

N/A

rx_gain

packet receive gain (dBM)

increases the sound level for receive packets

N/A

out_of_band_DTMF

out of band DTMF tone enable/disable

introduces a slight fixed delay into the voice path when enabled, and allows the sending of DTMF digits outside of the normal voice stream

N/A

Leave a Reply

Your email address will not be published. Required fields are marked *