Odd One-way Audio Problem (Mike Diehl)
Hi Mike
In rtp.conf, what are the port ranges you specify?
I had almost exactly the same problem not too long ago. People will phone, and sometimes it will work, sometimes not – one way audio would happen, then start working, then stop working.
The problem turned out to be that the port specification for RTP traffic in /etc/asterisk/rtp.conf was too wide.
It was set to
rtpstart=10000
rtpend=65535
(apparently by a previous maintainer / technician who worked on the system.)
The high port number was too high, and only after I investigated in detail with our trunk provider, were they able to determine that somtimes the Asterisk on my side was negotiating too high port numbers for RTP with their system.
I changed rtp.conf to read
rtpstart=10000
rtpend=20000
and all the random one-way audio problems have been gone for more than two months. This client now has had thousads of successful calls so far after this change was made.
I also had the issue where MOST calls in their office was fine (with rtp.conf at 10000 to 65535) though some would still fail, I’m guessing that was due to NATing not being done in the office (e. g. a wider “range” of RTP ports worked) vs. when they connected to their provider’s SIP trunk on the internet to negotiate calls where it was ignoring the higher ports (“too high” ports) or their local firewall wasn’t allowing some high ports to be opened that were “too high”.
Restricting the RTP port range between 10000 and 20000 in this case solved their problem definitively and forever.
E. g. something similar given that you start that “most of the time” things worked fine – which is exactly the symptom I had with this client.
Just a thought…
Regards
Stefan
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Hi all,
I have a user who is reporting one-way audio, but only when a call is made to or from particular PSTN (cell) numbers.
Their phones are behind a NAT router and my server is on the open Internet.
Calls within their office sound fine. Calls to/from most numbers sound fine.
When they took their phones home, those same phone numbers still had problems.
So, I don’t think it’s their network. I’ve taken pcaps of both legs of example calls. On the provider-side, I see 2-way audio. On the client-side, I only hear one side.
Most of the time, though, their phones work correctly.
Any ideas where to look to fix this?
Thanks in advance.
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3 thoughts on - Odd One-way Audio Problem (Mike Diehl)
Hi, and thank you for your suggestion!
As it turns out, my server didn’t even HAVE an rtp.conf file… (No, I don’t know how that happened…)
So I created one with:
rtpstart000
rtpend 000
and reloaded chan_sip.
I hope that is sufficient. Or do I need to restart asterisk completely?
Anyway, my user tested later that day and they are still having problems….
Any other ideas?
Mike.
Mike,
In those pcaps, are you seeing the exact same RTP traffic between provider side and client side?
Hi Mike
Oh yes, this:
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Note that the call ID is much longer than in the column display, e. g. a visual call ID in the sip show channelstats display may be
31f867c50ce
but the full call ID is then
32d867a55cfb563b7f59da01de84dc6f@xxx.xxx.xxx.xxx:5060
The full ID can be obtained by typing the partial ID as displayed in the list in the Asterisk CLI and then hitting TAB on the keyboard.
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will only work if you go
sip show channel 31f867c50ce
and THEN press TAB on the keyboard.
Apologies.
Stefan
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