Info "RTP audio difference"

Asterisk Asterisk Support
1

Hi all,

I set a new sip trunk and everything works fine. I noticed that when I use it for outgoing call asterisk’s cli outputs following logs. What that means?

-- Called PJSIP/0471065789@eolo-0471098455
   > 0x7f32e801fa00 -- Strict RTP learning after remote address set to: 10.40.99.16:32504
-- PJSIP/eolo-0471098455-00000003 is making progress passing it to PJSIP/101-00000002
   > 0x7f32e8053f80 -- Strict RTP learning after remote address set to: 192.168.1.227:4004
   > 0x7f32e8053f80 -- Strict RTP switching to RTP target address 192.168.1.227:4004 as source
   > 0x7f32e801fa00 -- Strict RTP switching to RTP target address 10.40.99.16:32504 as source

(0x7f32e804d720) RTP audio difference is 1782488536 set mark
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002
(0x7f32e804d720) RTP audio difference is 959232680 set mark
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002
(0x7f32e804d720) RTP audio difference is 959232520 set mark
– PJSIP/eolo-0471098455-00000003 is making progress passing it to PJSIP/101-00000002
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002
(0x7f32e804d720) RTP audio difference is 959232648 set mark
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002
(0x7f32e804d720) RTP audio difference is 959232560 set mark
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002
(0x7f32e804d720) RTP audio difference is 959232560 set mark
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002
(0x7f32e804d720) RTP audio difference is 959232552 set mark
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002
(0x7f32e804d720) RTP audio difference is 959232560 set mark
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002
(0x7f32e804d720) RTP audio difference is 959232536 set mark
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002
(0x7f32e804d720) RTP audio difference is 959232544 set mark
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002
(0x7f32e804d720) RTP audio difference is 959232560 set mark
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002
(0x7f32e804d720) RTP audio difference is 959232560 set mark
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002
(0x7f32e804d720) RTP audio difference is 959232552 set mark
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002
(0x7f32e804d720) RTP audio difference is 959232560 set mark
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002
(0x7f32e804d720) RTP audio difference is 959232552 set mark
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002
(0x7f32e804d720) RTP audio difference is 959232568 set mark
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002
(0x7f32e804d720) RTP audio difference is 959232536 set mark
– PJSIP/eolo-0471098455-00000003 requested media update control 26, passing it to PJSIP/101-00000002

2

This is a debug level message. It’s not a good idea to enable debug unless you’ve been asked to or understand what is going on, as it can raise questions like this. The message itself happens when the predicted timestamp differs from the actual calculated one sufficiently.

Looking further into it would require looking at the actual RTP streams involved.

3

Hi, thank you for reply but I’am not totally agree with you.

RTP debug is set to “off”.

If I make an outgoing call using a different trunk group that logs does not shown.

They are displayed only with one, of two trunks

4

The message is a core debug message[1], not from “rtp set debug on”. Something to do with the RTP stream is causing it.

[1] asterisk/res/res_rtp_asterisk.c at 22 · asterisk/asterisk

5

Thanks for the clarification. I’ll look into it.