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author | Peter Wu <peter@lekensteyn.nl> | 2016-12-02 11:24:23 +0100 |
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committer | Peter Wu <peter@lekensteyn.nl> | 2016-12-02 11:24:23 +0100 |
commit | 3cdb9eab57dac8497d13e3a9b90cf40505db767f (patch) | |
tree | b34da1c9d2e80d998efbea17c741820df9726f88 /sipsim | |
parent | 6951d901827e9d31a8c09a3a30d6d65ac25514c7 (diff) | |
download | wireshark-notes-3cdb9eab57dac8497d13e3a9b90cf40505db767f.tar.gz |
Added SIPp scenario and list of codecs supported by FS
Requires appropriately configured FreeSWITCH server that responds to a
call to sip:test@host by playing a fragment, then hanging up.
SIPp scenario was used to create a bunch of captures, uploaded to
https://wiki.wireshark.org/SampleCaptures#SIP_and_RTP
Diffstat (limited to 'sipsim')
-rw-r--r-- | sipsim/codecs.txt | 57 | ||||
-rw-r--r-- | sipsim/uac_media.xml | 92 |
2 files changed, 149 insertions, 0 deletions
diff --git a/sipsim/codecs.txt b/sipsim/codecs.txt new file mode 100644 index 0000000..94cac76 --- /dev/null +++ b/sipsim/codecs.txt @@ -0,0 +1,57 @@ + +# Note: codecname[@8000h|16000h|32000h[@XXi]] + +# mod_g723_1 is for passthrough +# G723/8000 4 + +# (in core) +#PCMU/8000 0 +#PCMA/8000 8 +# RFC 3555 valid sample sizes for L16: 8000, 11025, 16000, 22050, 24000, 32000, 44100, and 48000 +# L16 RTP clock rate is always 44100. (pt 10 and 11, but dynamic for others) +#L16/8000/2 +#L16/16000/2 +#L16/11025 +#L16/48000 + +# mod_com_g728 +# G728/8000 15 + +# mod_com_g729 +# G729/8000 18 + + +# (mostly) dynamic types +# mod_ilbc +#iLBC/8000 # note: a=fmtp:97 mode=20 or mode=30 + +# mod_siren +#G7221/16000 # note: a=fmtp:121 bitrate=24000 +#G7221/32000 # note: a=fmtp:122 bitrate=48000 + +# note, additionally also supported: PCMU/PCMA (G711) +# mod_spandsp +#GSM/8000 3 +#DVI4/8000 5 +#DVI4/16000 6 +#LPC/8000 7 +# G722/8000 on wire, but actual sample size is 16kHz +# G722/8000 9 +#G722/16000 +#G726-16/8000 +#G726-24/8000 +#G726-32/8000 +#G726-40/8000 +#AAL2-G726-16/8000 +#AAL2-G726-24/8000 +#AAL2-G726-32/8000 +#AAL2-G726-40/8000 + +# Maybe FS wants capital OPUS and SPEEX... +# mod_opus +#opus/48000/2 + +# (in core) +#speex/8000 +#speex/16000 +#speex/32000 diff --git a/sipsim/uac_media.xml b/sipsim/uac_media.xml new file mode 100644 index 0000000..51b5433 --- /dev/null +++ b/sipsim/uac_media.xml @@ -0,0 +1,92 @@ +<?xml version="1.0" encoding="ISO-8859-1" ?> +<!DOCTYPE scenario SYSTEM "sipp.dtd"> + +<!-- This program is free software; you can redistribute it and/or --> +<!-- modify it under the terms of the GNU General Public License as --> +<!-- published by the Free Software Foundation; either version 2 of the --> +<!-- License, or (at your option) any later version. --> +<!-- --> +<!-- This program is distributed in the hope that it will be useful, --> +<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --> +<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --> +<!-- GNU General Public License for more details. --> +<!-- --> +<!-- You should have received a copy of the GNU General Public License --> +<!-- along with this program; if not, write to the --> +<!-- Free Software Foundation, Inc., --> +<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --> +<!-- --> +<!-- Based on sipp default 'uac' scenario. --> +<!-- To make one call to test@10.0.2.15 from IP 10.0.2.20 using PCMU codec: + ./sipp -sf uac_media.xml 10.0.2.15 -i 10.0.2.20 -bind_local -m 1 -s test -key codec PCMU/8000 -key pt 99 + --> +<!-- --> + +<scenario name="Codec test"> + <send retrans="500"> + <![CDATA[ + INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 + Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] + From: "[codec]" <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] + To: [service] <sip:[service]@[remote_ip]:[remote_port]> + Call-ID: [call_id] + CSeq: 1 INVITE + Contact: sip:sipp@[local_ip]:[local_port] + Max-Forwards: 70 + Content-Type: application/sdp + Content-Length: [len] + + v=0 + o=- 42 42 IN IP[local_ip_type] [local_ip] + s=- + c=IN IP[media_ip_type] [media_ip] + t=0 0 + m=audio [media_port] RTP/AVP [pt] + a=rtpmap:[pt] [codec] + a=recvonly + ]]> + </send> + + <recv response="100" optional="true"> + </recv> + + <recv response="180" optional="true"> + </recv> + + <recv response="183" optional="true"> + </recv> + + <recv response="200" rtd="true"> + </recv> + + <send> + <![CDATA[ + ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 + Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] + From: "[codec]" <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] + To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] + Call-ID: [call_id] + CSeq: 1 ACK + Contact: sip:sipp@[local_ip]:[local_port] + Max-Forwards: 70 + Content-Length: 0 + ]]> + </send> + + <!-- wait for callee to hang up --> + <recv request="BYE"> + </recv> + + <send> + <![CDATA[ + SIP/2.0 200 OK + [last_Via:] + [last_From:] + [last_To:] + [last_Call-ID:] + [last_CSeq:] + Contact: <sip:[local_ip]:[local_port];transport=[transport]> + Content-Length: 0 + ]]> + </send> +</scenario> |