The setup:
FreePBX distro (same problem occurs with the 64 bits alpha-5.12.65-3/asterisk 12 and with the 64bits stable-5.211.65.6/asterisk11)
distro running on a server with fixed ip address, firewall disabled (all ports open). Server is on a 100Mbps guaranteed bandwidth connection, there is no packet loss and low ping between the input and output.
Input: DID number routed to the server IP address
Configuration: DISA on DID, then dial out the number entered
Output: SIP Voip Provider (premium quality)
Codec: G711-Alaw (input and output)
Problems:
- audio prompts, congestion messages & DISA dial tone are very choppy (timing issue?)
- when call is established, audio from source to destination is fine, low delay, good quality. Audio from destination to source is delayed by about 2 seconds, it’s very choppy. I noticed no audio packet is transmitted from destination to source if there is no audio or low level audio, there seems to be a 2 seconds delay (I need to keep talking more than 2 seconds nons top) in order to have audio packets comming in. Analysis shows average of 48 packets/sec on one leg (the good one) and 16/sec on the other (that means many missing packets).
What I have tried already:
- Checking connectivity
- Reinstall (tried 2 distro versions)
- Install on another server (same problem)
- add clock=pit in grub.conf
- Enable jitter buffer (didn’t help)
- Use other codecs (g711-ulaw, g729)
- Input with SIP instead of DID
- output directly to a SIP device instead of voip operator
- Pass through a sip gateway on input/output and forcing RTP relay (no improvement)
- tried “silenceSuppression=no” in asterisk SIP settings
- Checked logfiles for error or warnings
If anyone knows what could be causing this, that would really help me.
Thanks in advance.