signalwire
Freeswitch Bug when SDP on 180
Describe the bug Freeswitch send 183 instead of 180 ( with SDP ) when the 180 contains SDP
To Reproduce Steps to reproduce the behavior:
- create one sipp uac and sipp uas
- Dial from sipp uac to sipp uas through freeswitch
- sipp uas include SDP in 180
- FreeSWITCH forward 183 with SDP instead of 180 with SDP
Expected behavior Freeswitch must forward the SIP 180 along with SDP
Package version or git hash
- Version 1.10.6
Trace logs Provide freeswitch logs w/ DEBUG and UUID logging enabled
sipp uac
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<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: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[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
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=kamuac 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [rtpstream_audio_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
]]>
</send>
<recv response="100" rtd="true" optional="true">
</recv>
<recv response="180" rtd="true" optional="true">
</recv>
<recv response="183" rtd="true" optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true" rrs="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK [next_url] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[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]
[routes]
Max-Forwards: 70
Content-Length: 0
]]>
</send>
<nop>
<action>
<exec rtp_stream="/var/sipp/scripts/pcap/voice-man.wav,-1,8,PCMA/8000"/>
</action>
</nop>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<pause />
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500" crlf="true">
<![CDATA[
BYE [next_url] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
[routes]
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true" next="10">
</recv>
<label id="9" rtd="true" />
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[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>
<label id="10" rtd="true" />
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
sipp uas If applicable, provide the full backtrace from the core file.
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic UAS responder">
<recv request="INVITE" crlf="true">
</recv>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=fsuas 87308505 1 IN IP[local_ip_type] [local_ip]
s=-
t=0 0
c=IN IP[media_ip_type] [media_ip]
m=audio [rtpstream_audio_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<recv request="ACK" crlf="true">
</recv>
<nop>
<action>
<exec rtp_stream="/var/sipp/scripts/pcap/voice-woman.wav,-1,8,PCMA/8000"/>
</action>
</nop>
<pause />
<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>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
Looks like it has been discussed here... My understanding is, FS is a B2BUA, not a SIP proxy... https://lists.freeswitch.org/pipermail/freeswitch-users/2016-August/121974.html
Hello, you may already found a solution but to share my experience, I faced the same issue when deploying freeswitch as an sbc to handle traffic from an FR carrier.
For some reason, FS automatically converts received 180+SDP to 183+SDP and 183 without SDP to 180. The given reason is debatable but my solution was to make a patch to remove this automatic conversion and keep the same received progress code as it was a hard requirement for the certification tests.
Hi Guys, Did you found the correction to fix this bug? I met the same bug and it is very urgent for me to fix it. Thanks you very much for your feedback Bests regards
