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Replaced SIPp -mp with -min_rtp_port #25

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Replaced SIPp -mp with -min_rtp_port #25

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7 changes: 4 additions & 3 deletions lib/python/asterisk/sipp.py
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Original file line number Diff line number Diff line change
Expand Up @@ -714,7 +714,7 @@ def __evaluate_scenario_results(result):
default_args[defarg] = ('%s/sipp/%s' % (
self.test_dir, default_args[defarg]))

if '-mp' not in default_args:
if '-min_rtp_port' not in default_args:
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This is not backwards compatible. Personally the SIPp I use isn't new enough to have this, so it would no longer work for me. I'm on Ubuntu 22.04, so I expect others would experience the same. Is the version shipping in distributions or has to be built manually? Is it possible to make this backwards compatible to support both based on detection of the version?

# Current SIPp correctly chooses an available port for audio, but
# unfortunately it then attempts to bind to the audio port + n for
# things like rtcp and video without first checking if those other
Expand All @@ -724,8 +724,9 @@ def __evaluate_scenario_results(result):
# ourselves, and make sure all associated ports are available.
#
# num = 4 = ports for audio rtp/rtcp and video rtp/rtcp
default_args['-mp'] = str(get_available_port(
default_args.get('-i'), num=4))
port = get_available_port(default_args.get('-i'), num=4)
default_args['-min_rtp_port'] = str(port)
default_args['-max_rtp_port'] = str(port + 3)

for (key, val) in default_args.items():
sipp_args.extend([key, val])
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2 changes: 1 addition & 1 deletion tests/channels/pjsip/dialplan_functions/pjsip_channel/test-config.yaml
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Original file line number Diff line number Diff line change
Expand Up @@ -21,7 +21,7 @@ test-object-config:
-
scenarios:
- { 'key-args': { 'scenario': 'uac-no-hangup.xml', '-p': '5062', '-i': '127.0.0.1',
'-s': 'alice', '-rsa': '127.0.0.1:5061', '-s': 'alice', '-mp': '9050'} }
'-s': 'alice', '-rsa': '127.0.0.1:5061', '-s': 'alice', '-min_rtp_port': '9050'} }


ami-config:
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2 changes: 1 addition & 1 deletion tests/channels/pjsip/non_negotiated_frame_SSRC_change/test-config.yaml
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Expand Up @@ -21,7 +21,7 @@ test-object-config:
-
scenarios:
- { 'key-args': { 'scenario': 'uas_asterisk.xml', '-i': '127.0.0.1', '-p': '5700'} }
- { 'key-args': { 'scenario': 'uac_g719_g711.xml', '-i': '127.0.0.1', '-p': '5061', '-s': '3200000000', '-d': '20000', '-mp': '6000'} }
- { 'key-args': { 'scenario': 'uac_g719_g711.xml', '-i': '127.0.0.1', '-p': '5061', '-s': '3200000000', '-d': '20000', '-min_rtp_port': '6000'} }

properties:
dependencies:
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2 changes: 1 addition & 1 deletion tests/channels/pjsip/rtp_ptime/non-transcode/run-test
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Original file line number Diff line number Diff line change
Expand Up @@ -28,7 +28,7 @@ SIPP_SCENARIOS = [
'scenario' : 'uas_asterisk.xml',
'-i' : '127.0.0.1',
'-p' : '5700',
'-mp' : '6300',
'-min_rtp_port' : '6300',
'-message_file' : sippB_logfile,
'-error_file' : sippB_errfile,
'-trace_msg' : '-trace_err',
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2 changes: 1 addition & 1 deletion tests/channels/pjsip/rtp_ptime/transcode/run-test
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Original file line number Diff line number Diff line change
Expand Up @@ -28,7 +28,7 @@ SIPP_SCENARIOS = [
'scenario' : 'uas_asterisk.xml',
'-i' : '127.0.0.1',
'-p' : '5700',
'-mp' : '6300',
'-min_rtp_port' : '6300',
'-message_file' : sippB_logfile,
'-error_file' : sippB_errfile,
'-trace_msg' : '-trace_err',
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4 changes: 2 additions & 2 deletions tests/codecs/opus/fec/jitterbuffer/adaptive/test-config.yaml
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Expand Up @@ -21,10 +21,10 @@ test-object-config:
test-iterations:
-
scenarios:
- { 'key-args': { 'scenario':'invite_recv.xml', '-p':'5062', '-mp':'16002' } }
- { 'key-args': { 'scenario':'invite_recv.xml', '-p':'5062', '-min_rtp_port':'16002' } }
# Specify the RTP port so the rtp analyzer knows where to
# listen for packets. The analyzer defaults to 6001.
- { 'key-args': { 'scenario':'invite.xml', '-p':'5061', '-mp':'16001' } }
- { 'key-args': { 'scenario':'invite.xml', '-p':'5061', '-min_rtp_port':'16001' } }

rtp-analyzer-config:
# Expected time difference for ulaw
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4 changes: 2 additions & 2 deletions tests/codecs/opus/fec/jitterbuffer/fixed/test-config.yaml
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Original file line number Diff line number Diff line change
Expand Up @@ -21,10 +21,10 @@ test-object-config:
test-iterations:
-
scenarios:
- { 'key-args': { 'scenario':'invite_recv.xml', '-p':'5062', '-mp':'16002' } }
- { 'key-args': { 'scenario':'invite_recv.xml', '-p':'5062', '-min_rtp_port':'16002' } }
# Specify the RTP port so the rtp analyzer knows where to
# listen for packets. The analyzer defaults to 6001.
- { 'key-args': { 'scenario':'invite.xml', '-p':'5061', '-mp':'16001' } }
- { 'key-args': { 'scenario':'invite.xml', '-p':'5061', '-min_rtp_port':'16001' } }

rtp-analyzer-config:
# Expected time difference for ulaw
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4 changes: 2 additions & 2 deletions tests/codecs/opus/fec/no_jitterbuffer/test-config.yaml
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Original file line number Diff line number Diff line change
Expand Up @@ -21,10 +21,10 @@ test-object-config:
test-iterations:
-
scenarios:
- { 'key-args': { 'scenario':'invite_recv.xml', '-p':'5062', '-mp':'16002' } }
- { 'key-args': { 'scenario':'invite_recv.xml', '-p':'5062', '-min_rtp_port':'16002' } }
# Specify the RTP port so the rtp analyzer knows where to
# listen for packets. The analyzer defaults to 6001.
- { 'key-args': { 'scenario':'invite.xml', '-p':'5061', '-mp':'16001' } }
- { 'key-args': { 'scenario':'invite.xml', '-p':'5061', '-min_rtp_port':'16001' } }

rtp-analyzer-config:
# Expected time difference for ulaw
Expand Down