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RTP/SRTP port rotation functionality #807

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10 changes: 8 additions & 2 deletions include/rtpstream.hpp
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Original file line number Diff line number Diff line change
Expand Up @@ -204,8 +204,14 @@ int rtpstream_new_call(rtpstream_callinfo_t *callinfo);
void rtpstream_end_call(rtpstream_callinfo_t *callinfo);
int rtpstream_shutdown(std::unordered_map<pthread_t, std::string>& threadIDs);

int rtpstream_get_local_audioport(rtpstream_callinfo_t *callinfo);
int rtpstream_get_local_videoport(rtpstream_callinfo_t *callinfo);
int rtpstream_get_local_uac_audioport(rtpstream_callinfo_t *callinfo);
int rtpstream_get_local_uac_videoport(rtpstream_callinfo_t *callinfo);
int rtpstream_rotate_local_uac_audioport(rtpstream_callinfo_t *callinfo, int newPort);
int rtpstream_rotate_local_uac_videoport(rtpstream_callinfo_t *callinfo, int newPort);
int rtpstream_get_local_uas_audioport(rtpstream_callinfo_t *callinfo);
int rtpstream_get_local_uas_videoport(rtpstream_callinfo_t *callinfo);
int rtpstream_rotate_local_uas_audioport(rtpstream_callinfo_t *callinfo, int newPort);
int rtpstream_rotate_local_uas_videoport(rtpstream_callinfo_t *callinfo, int newPort);
void rtpstream_set_remote(rtpstream_callinfo_t* callinfo, int ip_ver, const char* ip_addr,
int audio_port, int video_port);

Expand Down
21 changes: 15 additions & 6 deletions include/sipp.hpp
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Expand Up @@ -151,7 +151,6 @@ cmd messages are received */
#define DEFAULT_RATE_PERIOD_MS 1000
#define DEFAULT_TRANSPORT T_UDP
#define DEFAULT_PORT 5060
#define DEFAULT_MEDIA_PORT 6000
#define DEFAULT_3PCC_PORT 6060
#define DEFAULT_SERVICE "service"
#define DEFAULT_AUTH_PASSWORD "password"
Expand All @@ -172,8 +171,12 @@ cmd messages are received */

#define DEFAULT_BEHAVIOR_ALL (DEFAULT_BEHAVIOR_BYE | DEFAULT_BEHAVIOR_ABORTUNEXP | DEFAULT_BEHAVIOR_PINGREPLY | DEFAULT_BEHAVIOR_BADCSEQ)

#define DEFAULT_MIN_RTP_PORT DEFAULT_MEDIA_PORT
#define DEFAULT_MAX_RTP_PORT 65535
#define DEFAULT_MIN_RTP_AUDIO_PORT 20000
#define DEFAULT_MAX_RTP_AUDIO_PORT 20998
#define DEFAULT_MIN_RTP_VIDEO_PORT 30000
#define DEFAULT_MAX_RTP_VIDEO_PORT 30998
#define DEFAULT_MIN_UDP_IMAGE_PORT 40000
#define DEFAULT_MAX_UDP_IMAGE_PORT 40998
#define DEFAULT_RTP_PAYLOAD 8
#define DEFAULT_RTP_THREADTASKS 20

Expand Down Expand Up @@ -251,8 +254,12 @@ MAYBE_EXTERN int control_port DEFVAL(0);
MAYBE_EXTERN int buff_size DEFVAL(65536);
MAYBE_EXTERN int tcp_readsize DEFVAL(65536);
MAYBE_EXTERN int hasMedia DEFVAL(0);
MAYBE_EXTERN int min_rtp_port DEFVAL(DEFAULT_MIN_RTP_PORT);
MAYBE_EXTERN int max_rtp_port DEFVAL(DEFAULT_MAX_RTP_PORT);
MAYBE_EXTERN int min_rtp_audio_port DEFVAL(DEFAULT_MIN_RTP_AUDIO_PORT);
MAYBE_EXTERN int max_rtp_audio_port DEFVAL(DEFAULT_MAX_RTP_AUDIO_PORT);
MAYBE_EXTERN int min_rtp_video_port DEFVAL(DEFAULT_MIN_RTP_VIDEO_PORT);
MAYBE_EXTERN int max_rtp_video_port DEFVAL(DEFAULT_MAX_RTP_VIDEO_PORT);
MAYBE_EXTERN int min_udp_image_port DEFVAL(DEFAULT_MIN_UDP_IMAGE_PORT);
MAYBE_EXTERN int max_udp_image_port DEFVAL(DEFAULT_MAX_UDP_IMAGE_PORT);
MAYBE_EXTERN int rtp_default_payload DEFVAL(DEFAULT_RTP_PAYLOAD);
MAYBE_EXTERN int rtp_tasks_per_thread DEFVAL(DEFAULT_RTP_THREADTASKS);
MAYBE_EXTERN int rtp_buffsize DEFVAL(65536);
Expand All @@ -265,7 +272,9 @@ MAYBE_EXTERN double videotolerance DEFVAL(1.0);

MAYBE_EXTERN bool rtp_echo_enabled DEFVAL(0);
MAYBE_EXTERN char media_ip[127]; /* also used for hostnames */
MAYBE_EXTERN int media_port DEFVAL(0);
MAYBE_EXTERN int media_audio_port DEFVAL(0);
MAYBE_EXTERN int media_video_port DEFVAL(0);
MAYBE_EXTERN int media_image_port DEFVAL(0);
MAYBE_EXTERN size_t media_bufsize DEFVAL(2048);
MAYBE_EXTERN bool media_ip_is_ipv6 DEFVAL(false);
MAYBE_EXTERN char remote_ip[127]; /* also used for hostnames */
Expand Down
2 changes: 1 addition & 1 deletion regress/github-#0196/run
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Original file line number Diff line number Diff line change
Expand Up @@ -16,7 +16,7 @@ udplisten_job=$!
trap "kill -9 $udplisten_job 2>/dev/null" EXIT

sippbg -sf uas.xml -p 5070 -key custom_media_port $uas_media_port
sippfg -m 1 -sf uac.xml 127.0.0.1:5070 -min_rtp_port $uac_media_port \
sippfg -m 1 -sf uac.xml 127.0.0.1:5070 -min_rtp_audio_port $uac_media_port \
-timeout 5 -timeout_error >/dev/null 2>&1
status=$?

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2 changes: 1 addition & 1 deletion regress/github-#0276/run
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Original file line number Diff line number Diff line change
Expand Up @@ -8,7 +8,7 @@ udplisten 6002 >tmp.log 2>&1 &
listenjob=$!

sippbg -sn uas -i 127.0.0.1 -p 5070 -m 1 \
-min_rtp_port 6200 # media port at 6200
-min_rtp_audio_port 6200 # audio media port at 6200
sippfg -sn uac -i 127.0.0.1 -m 1 127.0.0.1:5070 >uac.log 2>&1
fgok=$?

Expand Down
4 changes: 2 additions & 2 deletions sipp_scenarios/pfca_uac_apattern_crypto_simple_renegotiation.xml
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Original file line number Diff line number Diff line change
Expand Up @@ -31,9 +31,9 @@
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [rtpstream_audio_port] RTP/AVP 0 18 9 103 8 101
a=rtcp:[rtpstream_audio_port+1]
a=crypto:[cryptotag1audio] [cryptosuiteaescm128sha1801audio] inline:[cryptokeyparams1audio]
a=crypto:[cryptotag2audio] [cryptosuiteaescm128sha1322audio] inline:[cryptokeyparams2audio]
a=rtcp:[rtpstream_audio_port+1]
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
Expand Down Expand Up @@ -137,9 +137,9 @@
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [rtpstream_audio_port] RTP/AVP 0 18 9 103 8 101
a=rtcp:[rtpstream_audio_port+1]
a=crypto:[cryptotag1audio] [cryptosuiteaescm128sha1801audio] inline:[cryptokeyparams1audio]
a=crypto:[cryptotag2audio] [cryptosuiteaescm128sha1322audio] inline:[cryptokeyparams2audio]
a=rtcp:[rtpstream_audio_port+1]
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
Expand Down
215 changes: 215 additions & 0 deletions sipp_scenarios/pfca_uac_apattern_crypto_simple_renegotiation_rotation.xml
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@@ -0,0 +1,215 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<!-- UC360 INVITE/200/ACK/BYE/200 sample UAC scenario -->

<scenario name="Basic UC360 UAC">

<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: 16001 <sip:16001@[remote_ip]:[remote_port]>;tag=[call_number]
To: <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 10 INVITE
Contact: <sip:16001@[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: VIRTUAL Mitel-UC-Endpoint (Mitel UC360 Collaboration Point/2.1.0.99; 08:00:0F:74:80:E1)
Subject: Conference
Session-Expires: 3600;refresher=uas
Min-SE: 90
Supported: 100rel
Require: 100rel
Content-Length: [len]

v=0
o=16001 0 0 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 0 18 9 103 8 101
a=rtcp:[rtpstream_audio_port+1]
a=crypto:[cryptotag1audio] [cryptosuiteaescm128sha1801audio] inline:[cryptokeyparams1audio]
a=crypto:[cryptotag2audio] [cryptosuiteaescm128sha1322audio] inline:[cryptokeyparams2audio]
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/16000
a=fmtp:9 bitrate=64000
a=rtpmap:103 G7221/16000
a=fmtp:103 bitrate=32000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16

]]>
</send>

<recv response="100"
optional="true">
</recv>

<recv response="180">
<action>
<!-- NOTE: Save the 180 Ringing's CSeq header for later reuse -->
<ereg regexp=".*" search_in="hdr" header="CSeq:" check_it="true" assign_to="1" />
<!-- NOTE: Save the 180 Ringing's RSeq header for later reuse -->
<ereg regexp=".*" search_in="hdr" header="RSeq:" check_it="true" assign_to="2" />
</action>
</recv>

<!-- Send PRACK for 180 Ringing -->
<send retrans="500">
<![CDATA[

PRACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: 16001 <sip:16001@[local_ip]:[local_port]>;tag=[call_number]
To: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 11 PRACK
RAck: [$2][$1]
Contact: <sip:16001@[local_ip]:[local_port];transport=[transport]>
Max-Forwards: 70
Subject: Conference
Content-Length: 0

]]>
</send>

<!-- receive 200 OK / PRACK (180 Ringing) -->
<recv response="200">
</recv>

<!-- receive 200 OK / INVITE -->
<recv response="200">
</recv>

<!-- NOTE: [branch-5] is used to specify reuse of same [branch] value that was used five messages earlier (e.g. INVITE) -->
<send>
<![CDATA[

ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-5]
From: "16001" <sip:16001@[remote_ip]:[remote_port]>;tag=[call_number]
To: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 10 ACK
Content-Length: 0

]]>
</send>

<nop>
<action>
<exec rtp_stream="apattern,1,0,PCMU/8000" />
</action>
</nop>

<pause milliseconds="2000" />

<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: 16001 <sip:16001@[remote_ip]:[remote_port]>;tag=[call_number]
To: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 12 INVITE
Contact: <sip:16001@[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: VIRTUAL Mitel-UC-Endpoint (Mitel UC360 Collaboration Point/2.1.0.99; 08:00:0F:74:80:E1)
Subject: Conference
Session-Expires: 3600;refresher=uas
Min-SE: 90
Supported: 100rel
Require: 100rel
Content-Length: [len]

v=0
o=16001 0 1 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [rtpstream_audio_port+2] RTP/AVP 0 18 9 103 8 101
a=rtcp:[rtpstream_audio_port+1]
a=crypto:[cryptotag1audio] [cryptosuiteaescm128sha1801audio] inline:[cryptokeyparams1audio]
a=crypto:[cryptotag2audio] [cryptosuiteaescm128sha1322audio] inline:[cryptokeyparams2audio]
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/16000
a=fmtp:9 bitrate=64000
a=rtpmap:103 G7221/16000
a=fmtp:103 bitrate=32000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16

]]>
</send>

<recv response="100"
optional="true">
</recv>

<!-- receive 200 OK / INVITE -->
<recv response="200">
</recv>

<!-- NOTE: [branch-3] is used to specify reuse of same [branch] value that was used three messages earlier (e.g. INVITE) -->
<send>
<![CDATA[

ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
From: "16001" <sip:16001@[remote_ip]:[remote_port]>;tag=[call_number]
To: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 12 ACK
Content-Length: 0

]]>
</send>

<nop>
<action>
<exec rtp_stream="apattern,2,0,PCMU/8000" />
</action>
</nop>

<pause milliseconds="2000" />

<send retrans="500">
<![CDATA[

BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "16001" <sip:16001@[remote_ip]:[remote_port]>;tag=[call_number]
To: <sip:[service]@[remote_ip]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 13 BYE
Contact: <sip:16001@[local_ip]:[local_port];transport=[transport]>
Max-Forwards: 70
Subject: Conference
User-Agent: VIRTUAL Mitel-UC-Endpoint (Mitel UC360 Collaboration Point/2.1.0.99; 08:00:0F:74:80:E1)
Content-Length: 0

]]>
</send>

<recv response="200">
</recv>

<!-- 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>
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