[asterisk-bugs] [JIRA] (ASTERISK-24193) chan_sip option rtpkeepalive results in comfort noise packets being sent despite flowing RTP - resulting in audible interruptions to audio

[Rusty Newton (JIRA)]

Rusty Newton created ASTERISK-24193:
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             Summary: chan_sip option rtpkeepalive results in comfort noise packets being sent despite flowing RTP - resulting in audible interruptions to audio
                 Key: ASTERISK-24193
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24193
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Resources/res_rtp_asterisk
    Affects Versions: 11.11.0
            Reporter: alexr1
            Assignee: Rusty Newton
            Severity: Minor


I was experiencing minor audio clipping/interruptions and when I did a packet capture I found an RTP Comfort Noise packet corresponding with each interruption (PT=comfort noise). An easy fix is to disable rtpkeepalive.

In sip.conf, rtpkeepalive specifies the number of seconds of no rtp activity before sending a comfort noise rtp packet as a keep alive. In some cases, Asterisk sends one in the middle of rtp streams (It seems to send it to both parties simultaneously, too).

directmedia=no, so all rtp traffic is being handled by both asterisk servers.

Interruptions every 10 seconds:
AST11 Playing MOH <alaw> AST11 <alaw> SIP Phone

No Interruptions when transcoding takes place:
AST11 Playing MOH <alaw> AST11 <ulaw> SIP Phone
AST11 Playing MOH <ulaw> AST11 <alaw> SIP Phone

Unfortunately I don't have time to help further isolate the issue on our production system - mainly posting this so that someone else can find it if they're looking for a solution!



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