<div dir="ltr"><div>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.</div><div><br></div><div>Does that help anyone make sense of what's happening?</div><div><br></div><div>We have DAHDI running on the server:</div><div><br></div><div># asterisk -rx 'dahdi show version'<br>DAHDI Version: 3.0.0 Echo Canceller: <br># asterisk -rx 'dahdi show status' <br>Description Alarms IRQ bpviol CRC Fra Codi Options LBO</div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Thu, 19 May 2022 at 15:51, David Cunningham <<a href="mailto:dcunningham@voisonics.com" target="_blank">dcunningham@voisonics.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div>Hello,</div><div><br></div><div>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.</div><div><br></div><div>It can be reproduced with a very simple extension like this:</div><div>exten => 4081234567, 2, Dial(SIP/6501234567@bb.bb.bb.138)</div><div>where 4081234567 is the number we receive the call on, and 6501234567 is the number we're forwarding it out to.</div><div><br></div><div>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.</div><div><br></div><div>Would anyone have any ideas? Historically Asterisk didn't generate comfort noise - has that changed in version 13?<br></div><div><br>[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)<br>[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)<br>[May 17 20:26:24] DEBUG[17725][C-00000027] res_rtp_asterisk.c: Ooh, format changed from none to ulaw<br>[May 17 20:26:24] DEBUG[17725][C-00000027] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x14f4cc025998'<br>[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)<br>[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)</div><div><br></div><div>Thanks very much,<br></div><div><br>-- <br><div dir="ltr"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div>David Cunningham, Voisonics Limited<br><a href="http://voisonics.com/" target="_blank">http://voisonics.com/</a><br>USA: +1 213 221 1092<br>New Zealand: +64 (0)28 2558 3782</div></div></div></div></div></div></div></div></div></div></div></div></div>
</blockquote></div><br clear="all"><br>-- <br><div dir="ltr"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div>David Cunningham, Voisonics Limited<br><a href="http://voisonics.com/" target="_blank">http://voisonics.com/</a><br>USA: +1 213 221 1092<br>New Zealand: +64 (0)28 2558 3782</div></div></div></div></div></div></div></div></div></div></div>