Viewing Article

Full StarFull StarFull StarFull StarFull Star | Jan 7 2011, 2:44 AM |
Poor call quality on VoIP

You may get problems that audio sounds broken, this is generally caused by out of sequence packets.

With most systems this can be fixed by using either of these methods:

1) Enable Re-sequencing

Re-sequencing allows a little more time for packets to arrive. Please contact our support department to request we enable this feature.

2) Enable a Jitter Buffer

With FreePBX (Asterisk based Distro's like Trixbox, Elastix etc) you can change this in the sip_general_custom.conf

here are the variables you will require:

;Enable Jitter Settings
jitterbuffers=4
jbenable=yes
jbforce=yes
jbimpl=fixed
;jbimpl=adaptive
jbmaxsize=50
jbresyncthreshold=100
jblog=yes

Asterisk sip jbenable = yes|no
Enables the use of a jitterbuffer on the receiving side of a SIP channel. (Added in Version 1.4)

Asterisk sip jbforce = yes|no
Forces the use of a jitterbuffer on the receive side of a SIP channel. Defaults to "no". (Added in Version 1.4)

Asterisk sip jbmaxsize = Number
Max length of the jitterbuffer in milliseconds. (Added in Version 1.4)

Asterisk sip jbresyncthreshold = Number
Jump in the frame timestamps over which the jitterbuffer is resynchronized. Useful to improve the quality of the voice, with big jumps in/broken timestamps, usually sent from exotic devices and programs. Defaults to 1000. (Added in Version 1.4)

Asterisk sip jbimpl = fixed|adaptive
Jitterbuffer implementation, used on the receiving side of a SIP channel. Two implementations are currently available - "fixed" (with size always equals to jbmaxsize) and "adaptive" (with variable size). Defaults to fixed. (Added in Version 1.4)

Asterisk sip jblog = no|yes
Enables jitterbuffer frame logging. Defaults to "no". (Added in Version 1.4)

3) Utilise G729 Codec

The codec that you are using may be intolerant (G711 being a known example) of the disordered packets that are inevitable when using Sharedband. Please arrange for a more tolerant codec to be employed within your VOIP system. A known example of a more tolerant codec is G729.

4) Apply Line Prioritisation

Specific traffic can be prioritised by Sharedband using its 'Line Prioritisation' feature within these paramaters:

Priorised traffic must be directed thorough a single connection, this connection must be either node 1 or 2 and this connection must be designated

Prioritised capacity cannot exceed 95% of the total capacity available to this designated connection

To enable Line Prioritisation on the service, please provide the following values to us:

  • Which connection you would like to designate for prioritised traffic
  • What percentage of that connection's total capacity you would like to allocate for upstream prioritised traffic
  • What percentage of that connection's total capacity you would like to allocate for downstream prioritised traffic
  • Either the TOS, DSCP, or DS value of the traffic to be prioritised along with clarification of which of these values it is that you are providing