I’m trying to test one asterisk box by using Sipp application, the scenario is very simple:
/usr/local/bin/sipp -sf asterisk.xml -d 1000 -recv_timeout 10000 -m 50 -l 10
which sends 10 concurrent calls until 50 calls. Dialplan is simple as well, just answering the call and playback a file for 30 seconds then hung up.
Sipp showns a very bad statistic:
Successful call | 0 | 39
Failed call | 0 | 11
which is very strange for this capacity. The server is a VM, with this configuration:
$ free -m
total used free shared buff/cache available
Mem: 7981 512 4076 105 3393 6995
Swap: 4095 0 4095
OS: CentOS 7 X64
load average: 0.00, 0.02, 0.13
In wireshark/sngrep, i observed that packets are coming to Asterisk, but for some INVITES he is not replying back with 100 Trying and Sipp is trying to retransmit the packets until is getting timeouts.
In the Asterisk logs i didn’t found any errors and network statistic also, is not too bad:
$ watch -n1 “ss -4 -n -l | grep 5060”
udp UNCONN 0 0 *:5060 :
$ watch-n1 “ip -s link show”
ens224: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc mq state UP mode DEFAULT group default qlen 1000 link/ether 00:0c:29:fd:5f:6b brd ff:ff:ff:ff:ff:ff RX: bytes packets errors dropped overrun mcast 2565530245 12890391 0 0 0 0 TX: bytes packets errors dropped carrier collsns 1485871230 4254402 0 0 0 0
asterisk*CLI> core show settings
PBX Core settings ----------------- Version: 16.2.0 Build Options: BUILD_NATIVE, OPTIONAL_API, G711_NEW_ALGORITHM Maximum calls: 500 (Current 0) Maximum open file handles: 65000 Root console verbosity: 8 Current console verbosity: 8 Debug level: 0 Maximum load average: 0.000000 Minimum free memory: 0 MB Startup time: 10:28:37 Last reload time: 10:28:37 System: Linux/3.10.0-957.1.3.el7.x86_64 built by root on x86_64 2019-02-19 10:17:51 UTC System name: server1 Entity ID: 00:0c:29:fd:5f:61 PBX UUID: cc72507a-07ce-46d4-a1f8-7283604016fa Default language: en Language prefix: Enabled User name and group: asterisk/asterisk Executable includes: Enabled Transcode via SLIN: Enabled Transmit silence during rec: Disabled Generic PLC: Enabled Generic PLC on equal codecs: Disabled Min DTMF duration:: 80 Cache media frames: Enabled RTP use dynamic payloads: 1 RTP dynamic payload types: 35-63,96-127 * Subsystems ------------- Manager (AMI): Enabled Web Manager (AMI/HTTP): Disabled Call data records: Enabled Realtime Architecture (ARA): Enabled
Why the Asterisk cannot process all INVITES? where is the bottleneck?