Dovid Bender
2022-May-19 06:52 UTC
[asterisk-users] No audio for 10 seconds and then comfort noise
David,
Are you getting any RTP from the PSTN for either leg? If not it could be
that they assume you are behind NAT and want to see where the SRC of the
RTP before they send it back. We had a few carriers that did this. The
easiest way to get around it was to play a 0.5 second audio clip to the
incoming leg. This will send RTP to the inbound carrier, causing them to
send RTP back to you which would then hit the terminating carrier, which
then sends you back RTP completing the loop. The dialplan looks
something like this.
same => n, Progress()
same => n,
Playback(/var/lib//asterisk_custom/sounds/xc,noanswer)
same => n, Dial(SIP/+${EXTEN}@carrier,,)
On Thu, May 19, 2022 at 12:13 AM David Cunningham <dcunningham at
voisonics.com>
wrote:
> We found that the 10 seconds relates to the "rtpkeepalive =10" in
our
> sip.conf. If the rtpkeepalive is reduced then the delay reduces as well. If
> rtpkeepalive is removed from sip.conf then audio never starts flowing.
>
> Does that help anyone make sense of what's happening?
>
> We have DAHDI running on the server:
>
> # asterisk -rx 'dahdi show version'
> DAHDI Version: 3.0.0 Echo Canceller:
> # asterisk -rx 'dahdi show status'
> Description Alarms IRQ bpviol CRC Fra
> Codi Options LBO
>
>
> On Thu, 19 May 2022 at 15:51, David Cunningham <dcunningham at
voisonics.com>
> wrote:
>
>> Hello,
>>
>> We are running an Asterisk 13 server which is having a strange problem,
>> where on calls which are received from the PSTN and then forwarded out
to
>> the PSTN again there is no audio for the first 10 seconds of the call.
At
>> the 10 second mark audio starts flowing fine, and in a PCAP we see that
it
>> starts with a few "comfort noise" packers before the real
audio starts.
>>
>> It can be reproduced with a very simple extension like this:
>> exten => 4081234567, 2, Dial(SIP/6501234567 at bb.bb.bb.138)
>> where 4081234567 is the number we receive the call on, and 6501234567
is
>> the number we're forwarding it out to.
>>
>> In the Asterisk log we don't see any obvious reason for the audio
to
>> start flowing at the 10 second mark. All that is logged at that time is
the
>> following below.
>>
>> Would anyone have any ideas? Historically Asterisk didn't generate
>> comfort noise - has that changed in version 13?
>>
>> [May 17 20:26:24] VERBOSE[11933] res_rtp_asterisk.c: Sent Comfort Noise
>> RTP packet to aa.aa.aa.76:64280 (type 02, seq 009268, ts 000000, len
000001)
>> [May 17 20:26:24] VERBOSE[17794][C-00000027] res_rtp_asterisk.c: Got
RTP
>> packet from aa.aa.aa.76:64280 (type 00, seq 000662, ts 105920, len
000160)
>> [May 17 20:26:24] DEBUG[17725][C-00000027] res_rtp_asterisk.c: Ooh,
>> format changed from none to ulaw
>> [May 17 20:26:24] DEBUG[17725][C-00000027] res_rtp_asterisk.c: Starting
>> RTCP transmission on RTP instance '0x14f4cc025998'
>> [May 17 20:26:24] VERBOSE[17725][C-00000027] res_rtp_asterisk.c: Sent
RTP
>> packet to bb.bb.bb.20:35412 (type 00, seq 020934, ts 105920, len
000160)
>> [May 17 20:26:24] VERBOSE[17725][C-00000027] res_rtp_asterisk.c: Got
RTP
>> packet from bb.bb.bb.20:35412 (type 00, seq 029996, ts 102760, len
000160)
>>
>> Thanks very much,
>>
>> --
>> David Cunningham, Voisonics Limited
>> http://voisonics.com/
>> USA: +1 213 221 1092
>> New Zealand: +64 (0)28 2558 3782
>>
>
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
> --
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Joshua C. Colp
2022-May-19 08:38 UTC
[asterisk-users] No audio for 10 seconds and then comfort noise
On Thu, May 19, 2022 at 3:52 AM Dovid Bender <dovid at telecurve.com> wrote:> David, > > Are you getting any RTP from the PSTN for either leg? If not it could be > that they assume you are behind NAT and want to see where the SRC of the > RTP before they send it back. We had a few carriers that did this. The > easiest way to get around it was to play a 0.5 second audio clip to the > incoming leg. This will send RTP to the inbound carrier, causing them to > send RTP back to you which would then hit the terminating carrier, which > then sends you back RTP completing the loop. The dialplan looks > something like this. > > same => n, Progress() > same => n, > Playback(/var/lib//asterisk_custom/sounds/xc,noanswer) > same => n, Dial(SIP/+${EXTEN}@carrier,,) >I've also seen this happen due to networking equipment, specifically the equipment wanting Asterisk to send packets before allowing packets in. If both sides of the call are in this state, then you reach a stalemate and media won't flow. Since rtp_keepalive is generated by Asterisk, it gets sent, and media starts flowing. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20220519/be719e41/attachment.html>