[Asterisk-Users] Help with poor audio using SIP

[Robert Webb]

Hi all,

  I am having some audio quality issues with a provider 
under sip. The issue I am having is that the audio seems 
to be acting like a simplex connection. I have tested my 
setup with a second provider and the audio quality to them 
is great. Checked network type issues, latency, packet 
loss, etc. and all seems to be ok.

What I did find was a difference in the RTP debugs. Here 
is a capture from both providers:

RTP Debug from Teliax SIP connection w/ good audio:

Sent RTP packet to 208.139.204.228:10102 (type 0, seq 
9473, ts 135520, len 160)
Sent RTP packet to 208.139.204.228:10102 (type 0, seq 
9474, ts 135680, len 160)
Got RTP packet from 208.139.204.228:10102 (type 0, seq 
4467, ts 149600, len 160)
Got RTP packet from 208.139.204.228:10102 (type 0, seq 
4468, ts 149760, len 160)


RTP Debug from MPC connection w/ bad audio:

Sent RTP packet to 66.128.8.234:61414 (type 0, seq 3506, 
ts 51040, len 160)
Sent RTP packet to 66.128.8.234:61414 (type 0, seq 3507, 
ts 51200, len 160)
Got RTP packet from 66.128.8.234:61414 (type 0, seq 23701, 
ts 52480, len 80)
Got RTP packet from 66.128.8.234:61414 (type 0, seq 23702, 
ts 52560, len 80)
Got RTP packet from 66.128.8.234:61414 (type 0, seq 23703, 
ts 52640, len 80)
Got RTP packet from 66.128.8.234:61414 (type 0, seq 23704, 
ts 52720, len 80)


Notice that the lengths are different in the MPC packet 
capture. I am getting two packets from them to every one 
of mine. I was askied by them to set my packet size to 
20ms but do not know where to do that or if it can be 
done. They also stated that the packet size should be 
negotiated in the SIP INVITE and 200 OK messages.

Can someone point me in the right direction? Even just 
what to look for here.

I am currently running version 1.2.2, but had the same 
issues with 1.09 and 1.2.

Thanks,
Robert