I have noticed some occasional one way audio on a specific sub set of calls in my system. First, let me be more specific about what I mean about occasional one way audio. Unlike most of the posts I’ve seen (where the end fix was either NAT’ing or RTP issues) the calls in question will begin and progress just fine, and then in one direction (always the same direction) audio stops for a second or two. When this occurs, it seems like every other, or every third word is cut out for 5-15 seconds (estimate, I have not timed it), and then the call returns to normal. I experienced this probably 5 separate times during a 1.5 hour call.
A bit about the setup:
The inbound calls in question come in to an Asterisk system over an E1 line in Scotland, and are forwarded to another Asterisk system in the US over an IAX2 connection. The call is then delivered via SIP to either a handset (various models) or soft phone (various packages). After the call has progressed for some time, audio from me to the caller drops every other word for a few seconds.
On the Asterisk server with the E1 connection I see a fairly steady stream of the following error when watching the console (asterisk -cvvvr):
WARNING: chan_zap.c:6147 zt_pri_error: PRI: Read on 81 failed: Unknown error 500
NOTICE: chan_zap.c:6873 pri_dchannel: PRI got event: 8 on span 1
WARNING: chan_iax2.c:516 iax_error_output: Ignoring unknown information element ‘Unknown IE
The last of those is the most common.
I have checked the following:
Server/Network port speed/duplex match both sides allow the following codecs: ulaw (one side also has alaw/gsm)
jitterbuffer settings match each side has the other as an IAX friend, and registers to its peer
The UK server is running an old SVN version of Asterisk, and the US server is running 1.4.31, and performing an upgrade on either of these servers is not possible at this time.
Pleas advise on what else I can check to correct this issue.