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The spandspflow2pcap.py creates pcap files from fax.log files, generated through 'fax set debug on' when receiving a fax. An example fax.log is included as spandspflow2pcap.log. The sipp-sendfax.xml SIPp scenario can be used to replay that fax with a recent version of SIPp. ASTERISK-25660 #close Change-Id: I4de8f28b084055b482ab8a5b28d28b605b0ed526
332 lines
7.7 KiB
XML
332 lines
7.7 KiB
XML
<?xml version="1.0" encoding="UTF-8" ?>
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<!DOCTYPE scenario SYSTEM "sipp.dtd">
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<scenario name="INVITE (optional auth), re-INVITE to T38 and send a Fax, Walter Doekes 2012-2013">
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<!--
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NOTE: Creating a sipp-sendfax.pcap is as easy as:
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- receive a fax with asterisk
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- get the incoming side of the spansdp.log (use 'fax set debug on',
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check your logger.conf)
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- feed it to spandspflow2pcap.py
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NOTE: sipp-sendfax.xml requires image pcap play support in SIPp. This
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means a version above 3.5.0, or the master git branch from
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https://github.com/SIPp/sipp.
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-->
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<label id="invite"/>
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<send retrans="500" start_txn="invite">
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<![CDATA[
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INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
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Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
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From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
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To: sip:[tel]@[remote_ip]:[remote_port]
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Contact: sip:[service]@[local_ip]:[local_port]
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Call-ID: [call_id]
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CSeq: [cseq] INVITE
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Max-Forwards: 70
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Content-Type: application/sdp
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Content-Length: [len]
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v=0
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o=- 144969 144969 IN IP[local_ip_type] [local_ip]
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s=-
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c=IN IP[media_ip_type] [media_ip]
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t=0 0
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m=audio [media_port] RTP/AVP 8 0
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a=rtpmap:8 PCMA/8000
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a=rtpmap:0 PCMU/8000
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]]>
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</send>
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<recv response="100" optional="true" response_txn="invite"/>
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<recv response="180" optional="true" response_txn="invite"/>
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<recv response="181" optional="true" response_txn="invite"/>
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<recv response="183" optional="true" response_txn="invite"/>
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<recv response="200" optional="true" rrs="true" response_txn="invite" next="invite-ack"/>
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<recv response="401" optional="true" rrs="true" next="invite-with-auth" auth="true" rrs="true" response_txn="invite"/>
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<recv response="407" auth="true" rrs="true" response_txn="invite"/>
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<label id="invite-with-auth"/>
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<send ack_txn="invite">
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<![CDATA[
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ACK sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
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[last_Via:]
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[last_From:]
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[last_To:]
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Contact: sip:[service]@[local_ip]:[local_port]
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Call-ID: [call_id]
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CSeq: [cseq] ACK
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Max-Forwards: 70
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Content-Length: 0
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]]>
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</send>
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<send retrans="500" start_txn="invite">
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<![CDATA[
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INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
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Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
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From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
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To: sip:[tel]@[remote_ip]:[remote_port]
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Contact: sip:[service]@[local_ip]:[local_port]
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[authentication]
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Call-ID: [call_id]
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CSeq: [cseq] INVITE
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Max-Forwards: 70
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Content-Type: application/sdp
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Content-Length: [len]
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v=0
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o=- 144969 144969 IN IP[local_ip_type] [local_ip]
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s=-
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c=IN IP[media_ip_type] [media_ip]
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t=0 0
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m=audio [media_port] RTP/AVP 8 0
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a=rtpmap:8 PCMA/8000
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a=rtpmap:0 PCMU/8000
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]]>
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</send>
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<recv response="100" response_txn="invite"/>
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<recv response="180" optional="true" response_txn="invite"/>
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<recv response="181" optional="true" response_txn="invite"/>
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<recv response="183" optional="true" response_txn="invite"/>
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<recv response="200" rrs="true" response_txn="invite"/>
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<label id="invite-ack"/>
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<send ack_txn="invite">
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<![CDATA[
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ACK [next_url] SIP/2.0
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[last_Via:]
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[routes]
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[last_From:]
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[last_To:]
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Contact: sip:[service]@[local_ip]:[local_port]
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Call-ID: [call_id]
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CSeq: [cseq] ACK
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Max-Forwards: 70
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Content-Length: 0
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]]>
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</send>
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<!-- Send a bit of noise to start the RTP. You may need to use
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the -i MY_IP command line option. -->
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<pause milliseconds="500"/>
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<nop>
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<action>
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<exec play_pcap_audio="g711a.pcap"/>
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</action>
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</nop>
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<pause milliseconds="500"/>
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<!-- *****************************************************************
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Initiate re-INVITE to T38.
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****************************************************************** -->
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<send start_txn="t38invite">
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<![CDATA[
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INVITE [next_url] SIP/2.0
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Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
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[routes]
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[last_From:]
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[last_To:]
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Contact: sip:[service]@[local_ip]:[local_port]
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Call-ID: [call_id]
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CSeq: [cseq] INVITE
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Max-Forwards: 70
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Content-Type: application/sdp
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Content-Length: [len]
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v=0
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o=- 145280 145280 IN IP[local_ip_type] [local_ip]
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s=-
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c=IN IP[media_ip_type] [media_ip]
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t=0 0
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m=image [media_port] udptl t38
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a=T38FaxVersion:0
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a=T38MaxBitRate:14400
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a=T38FaxRateManagement:transferredTCF
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a=T38FaxMaxBuffer:200
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a=T38FaxMaxDatagram:200
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a=T38FaxUdpEC:t38UDPRedundancy
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]]>
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</send>
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<recv response="100" optional="true" response_txn="t38invite"/>
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<recv response="488" optional="true" response_txn="t38invite" next="abort"/>
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<recv response="200" response_txn="t38invite"/>
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<send ack_txn="t38invite">
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<![CDATA[
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ACK [next_url] SIP/2.0
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[last_Via:]
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[routes]
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[last_From:]
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[last_To:]
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Contact: sip:[service]@[local_ip]:[local_port]
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Call-ID: [call_id]
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CSeq: [cseq] ACK
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Max-Forwards: 70
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Content-Length: 0
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]]>
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</send>
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<!-- Enable the nop/action below to replay the FAX image. You may need to use
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the -i MY_IP command line option. -->
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<pause milliseconds="500"/>
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<nop>
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<action>
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<exec play_pcap_image="sipp-sendfax.pcap"/>
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</action>
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</nop>
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<!-- *****************************************************************
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Wait for re-INVITE back to audio.
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****************************************************************** -->
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<recv request="INVITE"/>
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<send retrans="500">
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<![CDATA[
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SIP/2.0 200 OK
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[last_Via:]
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[last_Record-Route:]
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[last_From:]
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[last_To:]
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Contact: sip:[service]@[local_ip]:[local_port]
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[last_Call-ID:]
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[last_CSeq:]
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Max-Forwards: 70
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Content-Type: application/sdp
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Content-Length: [len]
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v=0
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o=- 146312 146312 IN IP[local_ip_type] [local_ip]
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s=-
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c=IN IP[media_ip_type] [media_ip]
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t=0 0
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m=audio [media_port] RTP/AVP 8 0
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a=rtpmap:8 PCMA/8000
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a=rtpmap:0 PCMU/8000
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]]>
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</send>
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<recv request="ACK"/>
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<recv request="BYE"/>
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<send next="done">
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<![CDATA[
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SIP/2.0 200 OK
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[last_Via:]
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[last_Record-Route:]
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[last_From:]
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[last_To:]
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Contact: sip:[service]@[local_ip]:[local_port]
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[last_Call-ID:]
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[last_CSeq:]
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Content-Length: 0
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]]>
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</send>
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<!-- *****************************************************************
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Abort the call ourselves
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****************************************************************** -->
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<label id="abort"/>
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<send ack_txn="t38invite">
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<![CDATA[
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ACK [next_url] SIP/2.0
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[last_Via:]
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[routes]
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[last_From:]
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[last_To:]
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Contact: sip:[service]@[local_ip]:[local_port]
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Call-ID: [call_id]
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CSeq: [cseq] ACK
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Max-Forwards: 70
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Content-Length: 0
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]]>
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</send>
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<send start_txn="bye">
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<![CDATA[
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BYE [next_url] SIP/2.0
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[last_Via:]
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[routes]
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[last_From:]
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[last_To:]
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Contact: sip:[service]@[local_ip]:[local_port]
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Call-ID: [call_id]
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CSeq: [cseq] BYE
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Max-Forwards: 70
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Content-Length: 0
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]]>
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</send>
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<recv response="200" response_txn="bye" next="done"/>
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<!-- *****************************************************************
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Finalize
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****************************************************************** -->
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<label id="done"/>
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<!-- Keep call open to be able to retransmit stuff -->
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<timewait milliseconds="2000"/>
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</scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->
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