Call is disconnected when unhold

Asterisk Asterisk SIP
1

I use Asterisk18.
Call is disconnected when unhold.

pjsip settings

  • rtp_timeout : 30
  • rtp_timeout_hold: 600

When the call was on hold for more than rtp_timeout, the call was disconnected if the arrival of rtp was delayed even by 1 second after releasing the hold.
Can anyone help us?

[May 22 14:56:41] VERBOSE[3392051] res_pjsip_logger.c: <--- Received SIP request (1301 bytes) from TLS:xxx.xxx.xxx.xxx:xxx --->
INVITE sips:asterisk@yyyy.yyyy.yyyy.yyyy:yyyy;transport=TLS SIP/2.0

a=inactive

[May 22 14:56:41] VERBOSE[1297986] res_pjsip_logger.c: <--- Transmitting SIP response (986 bytes) to TLS:xxx.xxx.xxx.xxx:xxx --->
SIP/2.0 200 OK

a=inactive

[May 22 14:56:41] VERBOSE[3992256][C-00012788] res_musiconhold.c: Started music on hold, class 'default', on channel 'PJSIP/zzzzz-00032781'

[May 22 14:56:41] VERBOSE[3392051] res_pjsip_logger.c: <--- Received SIP request (463 bytes) from TLS:xxx.xxx.xxx.xxx:xxx --->
ACK sip:asterisk@yyyy.yyyy.yyyy.yyyy:yyyy;transport=TLS SIP/2.0


[May 22 14:57:24] VERBOSE[3392051] res_pjsip_logger.c: <--- Received SIP request (1300 bytes) from TLS:xxx.xxx.xxx.xxx:xxx --->
INVITE sip:asterisk@yyyy.yyyy.yyyy.yyyy:yyyy;transport=TLS SIP/2.0

a=sendrecv

[May 22 14:57:24] VERBOSE[3593248] res_pjsip_logger.c: <--- Transmitting SIP response (986 bytes) to TLS:xxx.xxx.xxx.xxx:xxx --->
SIP/2.0 200 OK

a=sendrecv


[May 22 14:57:24] VERBOSE[3992256][C-00012788] res_musiconhold.c: Stopped music on hold on PJSIP/zzzzz-00032781
-> ACK arrived late

[May 22 14:57:25] NOTICE[3392199] res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/1000-00032782' for lack of audio RTP activity in 44 seconds

[May 22 14:57:25] VERBOSE[3392051] res_pjsip_logger.c: <--- Transmitting SIP response (986 bytes) to TLS:xxx.xxx.xxx.xxx:xxx --->
SIP/2.0 200 OK

a=sendrecv

[May 22 14:57:25] VERBOSE[3392051] res_pjsip_logger.c: <--- Received SIP request (463 bytes) from TLS:xxx.xxx.xxx.xxx:xxx --->
ACK sip:asterisk@yyyy.yyyy.yyyy.yyyy:yyyy;transport=TLS SIP/2.0

[May 22 14:57:25] VERBOSE[3392051] res_pjsip_logger.c: <--- Received SIP request (463 bytes) from TLS:xxx.xxx.xxx.xxx:xxx --->
ACK sip:asterisk@yyyy.yyyy.yyyy.yyyy:yyyy;transport=TLS SIP/2.0

[May 22 14:57:25] VERBOSE[3617167] res_pjsip_logger.c: <--- Transmitting SIP request (535 bytes) to TLS:xxx.xxx.xxx.xxx:xxx --->
BYE sips:1000@xxx.xxx.xxx.xxx:xxx;transport=TLS;ob SIP/2.0

[May 22 14:57:25] VERBOSE[3392051] res_pjsip_logger.c: <--- Received SIP response (434 bytes) from TLS:xxx.xxx.xxx.xxx:xxx --->
SIP/2.0 200 OK
2

Disconnecting channel ‘PJSIP/1000-00032782’ for lack of audio RTP activity in 44 seconds

I’ve never seen that myself. Are there actually any packets? rtp set debug on should show what is going on.

3

thank you for your reply.
Receiving rtp packets has been stopped because the call is inactive and on hold.

4

Not even something like keep-alive bytes? I am not sure whether continuing a call triggers a complete renegotiation of the media.

5

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