Testing asterisk with SIPp UAC and UAS scenarios


Hi all,

We are using asterisk 15.5 with pjsip. To load test the system we are using SIPp. We were successfully able to place calls from sipp UAC scenario to asterisk. We are trying to forward the call to another sipp with UAS scenario to complete entire call flow scenario. We are able to see that asterisk is sending INVITE request with given UAS context but I am not seeing any requests in sipp, also we are not seeing any response to INVITE request in asterisk as well.

PJSIP configuration:

[8001] (UAS peer)
type = aor
max_contacts = 100
contact = sip: (listening on IP and port)

type = auth
username = 8001
password =

type = endpoint
context = phones
disallow = all
allow = gsm
allow = ulaw
allow = alaw
force_rport = yes
rewrite_contact = yes
auth = 8001
outbound_auth = 8001
aors = 8001

[8002](UAC peer)
type = aor
max_contacts = 100

;type = auth
;username = 8002
;password =

type = endpoint
context = sipp
disallow = all
allow = gsm
allow = ulaw
allow = alaw
;force_rport = yes
;rewrite_contact = yes
;auth = 8002
;outbound_auth = 8002
aors = 8002

SIPp commands:

UAC: placing the calls to asterisk: ./sipp -sn uac -d 120000 -s 8002 -l 1 -m 1

UAS: server to receive calls from asterisk: ./sipp -sn uas -s 8001 -p 8001 -t u1
(This is listening on 8001 port, where asterisk needs to send the traffic)

exten => 8002,1,NoOp()
same => n,Verbose(“Call from sipp UAC”)
same => n,Dial(PJSIP/8001)
same => n,Hangup()

Please let us know if there’s any mistake/missing in my asterisk configuration or sipp command.


You can figure out where the problem is by doing a tshark trace on the Asterisk server and on the SIPp UAS server. You should also be able to see in the SIPp UAS output if it is discarding received SIP messages, in which case the format of the incoming message does not match the SIPp UAS config file. The best way to resolve this though is by first using traces and seeing where the unexpected behavior is occurring.