Asterisk queue answer delay problem

Asterisk Asterisk Support
1

Hi there,

First, It’s about outbound call’s results. Connection was normal when answer to call. Caller received my voice in first 1 second. There is no problem at this point. it just to compare with queue.

Second, about that queue’s calls. Agent heard my voice from second second when received call from queue.

Thanks…

2

SIP or not? Channel driver type? NAT environment? Does peer use late offer SDP? Does peer provide true addresses in SDP? Logs showing SDP? RTP debug output showing timing of RTP events?

3

Duplicated as Freepbx queue answer delay problem - Applications / Modules - FreePBX Community Forums

4

Hi david551,
We use PJSIP, WebRTC and DTLS.

== Spawn extension (from-internal, 3000, 1) exited non-zero on 'PJSIP/3000-000006ab'
  -- PJSIP/3000-000006ab Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
== Begin MixMonitor Recording Local/PHONE_NUMBER@localdialer-0000069f;1
  -- Called PJSIP/3000/sip:3000@MY_IP:49788;transport=ws
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio CoS mark 5
== DTLS ECDH initialized (automatic), faster PFS enabled
     > 0x7fa95c053770 -- Strict RTP switching to RTP target address RTP_IP:12584 as source
     > 0x7fa95c053770 -- Strict RTP learning complete - Locking on source address RTP_IP:12584
  -- PJSIP/3000-000006ab is ringing
     > 0x7fa95c27ff60 -- Strict RTP learning after remote address set to: 192.168.1.36:43495
  -- PJSIP/3000-000006ab answered Local/PHONE_NUMBER@localdialer-0000069f;1
  -- Channel PJSIP/3000-000006ab joined 'simple_bridge' basic-bridge <243be73a-18be-48f7-8439-b342439b09b1>
  -- Channel Local/PHONE_NUMBER@localdialer-0000069f;1 joined 'simple_bridge' basic-bridge <243be73a-18be-48f7-8439-b342439b09b1>
     > 0x7fa95c27ff60 -- Strict RTP learning after ICE completion
     > 0x7fa95c27ff60 -- Strict RTP learning after remote address set to: MY_IP:43495
     > 0x7fa95c27ff60 -- Strict RTP switching to RTP target address MY_IP:43495 as source
  -- Channel PJSIP/3000-000006ab left 'simple_bridge' basic-bridge <243be73a-18be-48f7-8439-b342439b09b1>
  -- Channel Local/PHONE_NUMBER@localdialer-0000069f;1 left 'simple_bridge' basic-bridge <243be73a-18be-48f7-8439-b342439b09b1>
== Spawn extension (macro-dial-one, s, 56) exited non-zero on 'Local/PHONE_NUMBER@localdialer-0000069f;1' in macro 'dial-one'
== Spawn extension (macro-exten-vm, s, 18) exited non-zero on 'Local/PHONE_NUMBER@localdialer-0000069f;1' in macro 'exten-vm'
== Spawn extension (ext-local, 3000, 3) exited non-zero on 'Local/PHONE_NUMBER@localdialer-0000069f;1'
5

Do was helpful to understand about problem?

6

RTP late starts when incoming called but outgoing starts instant.

7

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.