I actually checked out the latest version of FreeSWITCH in the SVN repository.
I have the following configured in
/usr/local/freeswitch/conf/dialplan/default.xml:
<extension name="setup_media" continue="true">
<condition field="${sip_nat_detected}" expression="true">
<action application="set" data="proxy_media=true" />
<action application="set" data="bypass_media=false" />
<anti-action application="set" data="proxy_media=false" />
<anti-action application="set" data="bypass_media=true" />
</condition>
</extension>
I have the following configured in /usr/local/freeswitch/conf/vars.xml:
<X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G729,i...@20i,G722,PCMU,PCMA"/>
<X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=G729,i...@20i,G722,PCMU,PCMA"/>
Here is the SIP trace for the failing call:
Nov 23 17:55:05.245 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE
sip:+19725357...@ipipgw.ipdimensions.com:5060;user=phone;transport=UDP;maddr=168.75.202.246
SIP/2.0
v: SIP/2.0/UDP
65.211.120.237:5060;branch=z9hG4bKec920f9119165c414d2f6229bb6a76ac.8e1ce24
Record-Route: <sip:65.211.120.237:5060;lr>
v: SIP/2.0/UDP
63.77.76.236:5060;branch=z9hG4bK19a30c0f46372620ff158f019d0ce5df.24ee0396;received=63.77.76.236
record-route: <sip:63.77.76.236;lr>
f:
<sip:+19729831...@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3d9f0a-5460a3be-3d9f0a
t: <sip:+19725357...@63.77.76.236:5060;user=phone>
i: a14d9878d065adc713c43d9f0af0b542beb67295e9c2c7438-0569-6585
CSeq: 1 INVITE
Max-Forwards: 16
k: 100rel, replaces
allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
v: SIP/2.0/UDP
DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3d9f0a-f0b542be-62e0db38;received=199.173.101.208
m: <sip:199.173.101.208:5060;transport=UDP>
c: application/SDP
l: 210
P-Asserted-Identity: <sip:9729831...@63.77.76.236;user=phone>
Privacy: none
v=0
o=- 641026559 641026559 IN IP4 199.173.111.147
s=-
c=IN IP4 199.173.111.147
t=0 0
m=audio 33344 RTP/AVP 18 0 8 101
a=ptime:20
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 23 17:55:05.257 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
65.211.120.237:5060;branch=z9hG4bKec920f9119165c414d2f6229bb6a76ac.8e1ce24,SIP/2.0/UDP
63.77.76.236:5060;branch=z9hG4bK19a30c0f46372620ff158f019d0ce5df.24ee0396;received=63.77.76.236,SIP/2.0/UDP
DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3d9f0a-f0b542be-62e0db38;received=199.173.101.208
From:
<sip:+19729831...@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3d9f0a-5460a3be-3d9f0a
To: <sip:+19725357...@63.77.76.236:5060;user=phone>
Date: Mon, 23 Nov 2009 23:55:05 GMT
Call-ID: a14d9878d065adc713c43d9f0af0b542beb67295e9c2c7438-0569-6585
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Nov 23 17:55:05.257 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:19725357...@168.75.202.212:5062 SIP/2.0
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3
From: <sip:19729831...@168.75.202.246>;tag=105BD148-201C
To: <sip:19725357...@168.75.202.212>
Date: Mon, 23 Nov 2009 23:55:05 GMT
Call-ID: 74e5b003-d7c211de-a29ad9df-3419a...@168.75.202.246
Supported: timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1961129755-3619819998-2727664095-874095366
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1259020505
Contact: <sip:19729831...@168.75.202.246:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 15
P-Asserted-Identity: <sip:19729831...@168.75.202.246>
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 314
v=0
o=CiscoSystemsSIP-GW-UserAgent 5041 5861 IN IP4 168.75.202.246
s=SIP Call
c=IN IP4 199.173.111.147
t=0 0
m=audio 33344 RTP/AVP 18 0 8 101
c=IN IP4 199.173.111.147
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 23 17:55:05.261 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3
From: <sip:19729831...@168.75.202.246>;tag=105BD148-201C
To: <sip:19725357...@168.75.202.212>
Call-ID: 74e5b003-d7c211de-a29ad9df-3419a...@168.75.202.246
CSeq: 101 INVITE
Timestamp: 1259020505 0.000345
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M
Content-Length: 0
Nov 23 17:55:05.309 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3
From: <sip:19729831...@168.75.202.246>;tag=105BD148-201C
To: <sip:19725357...@168.75.202.212>;tag=DFKSy9Q5DK1Na
Call-ID: 74e5b003-d7c211de-a29ad9df-3419a...@168.75.202.246
CSeq: 101 INVITE
Contact: <sip:19725357...@168.75.202.212:5062;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Content-Length: 0
P-Asserted-Identity: "19725357722" <sip:19725357...@168.75.202.212>
Nov 23 17:55:05.309 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
65.211.120.237:5060;branch=z9hG4bKec920f9119165c414d2f6229bb6a76ac.8e1ce24,SIP/2.0/UDP
63.77.76.236:5060;branch=z9hG4bK19a30c0f46372620ff158f019d0ce5df.24ee0396;received=63.77.76.236,SIP/2.0/UDP
DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3d9f0a-f0b542be-62e0db38;received=199.173.101.208
From:
<sip:+19729831...@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3d9f0a-5460a3be-3d9f0a
To: <sip:+19725357...@63.77.76.236:5060;user=phone>;tag=105BD180-BD7
Date: Mon, 23 Nov 2009 23:55:05 GMT
Call-ID: a14d9878d065adc713c43d9f0af0b542beb67295e9c2c7438-0569-6585
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:19725357...@168.75.202.246:5060>
Record-Route: <sip:65.211.120.237:5060;lr>,<sip:63.77.76.236;lr>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Nov 23 17:55:08.397 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3
From: <sip:19729831...@168.75.202.246>;tag=105BD148-201C
To: <sip:19725357...@168.75.202.212>;tag=DFKSy9Q5DK1Na
Call-ID: 74e5b003-d7c211de-a29ad9df-3419a...@168.75.202.246
CSeq: 101 INVITE
Contact: <sip:19725357...@168.75.202.212:5062;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Min-SE: 1800
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 202
P-Asserted-Identity: "19725357722" <sip:19725357...@168.75.202.212>
v=0
o=- 211627 211627 IN IP4 192.168.1.4
s=-
c=IN IP4 173.57.44.212
t=0 0
m=audio 0 RTP/AVP 96 101
a=rtpmap:96 G729a/8000
a=fmtp:96 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 23 17:55:08.397 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:19725357...@168.75.202.212:5062;transport=udp SIP/2.0
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659B6A
From: <sip:19729831...@168.75.202.246>;tag=105BD148-201C
To: <sip:19725357...@168.75.202.212>;tag=DFKSy9Q5DK1Na
Date: Mon, 23 Nov 2009 23:55:05 GMT
Call-ID: 74e5b003-d7c211de-a29ad9df-3419a...@168.75.202.246
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 23 17:55:08.397 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:19725357...@168.75.202.212:5062;transport=udp SIP/2.0
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659C1B3D
From: <sip:19729831...@168.75.202.246>;tag=105BD148-201C
To: <sip:19725357...@168.75.202.212>;tag=DFKSy9Q5DK1Na
Date: Mon, 23 Nov 2009 23:55:05 GMT
Call-ID: 74e5b003-d7c211de-a29ad9df-3419a...@168.75.202.246
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
P-Asserted-Identity: <sip:19729831...@168.75.202.246>
Timestamp: 1259020508
CSeq: 102 BYE
Reason: Q.850;cause=65
Content-Length: 0
Nov 23 17:55:08.401 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659C1B3D
From: <sip:19729831...@168.75.202.246>;tag=105BD148-201C
To: <sip:19725357...@168.75.202.212>;tag=DFKSy9Q5DK1Na
Call-ID: 74e5b003-d7c211de-a29ad9df-3419a...@168.75.202.246
CSeq: 102 BYE
Timestamp: 1259020508 0.000093
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M
Content-Length: 0
Nov 23 17:55:08.401 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659C1B3D
From: <sip:19729831...@168.75.202.246>;tag=105BD148-201C
To: <sip:19725357...@168.75.202.212>;tag=DFKSy9Q5DK1Na
Call-ID: 74e5b003-d7c211de-a29ad9df-3419a...@168.75.202.246
CSeq: 102 BYE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Content-Length: 0
The SIP call actually fails.
If I remove the following from /usr/local/freeswitch/conf/dialplan/default.xml:
<extension name="setup_media" continue="true">
<condition field="${sip_nat_detected}" expression="true">
<action application="set" data="proxy_media=true" />
<action application="set" data="bypass_media=false" />
<anti-action application="set" data="proxy_media=false" />
<anti-action application="set" data="bypass_media=true" />
</condition>
</extension>
And I change this line in /usr/local/freeswitch/conf/vars.xml from
<X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G729,i...@20i,G722,PCMU,PCMA"/>
to
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=i...@20i,G722,PCMU,PCMA"/>
And I change this line in /usr/local/freeswitch/conf/vars.xml from
<X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=G729,i...@20i,G722,PCMU,PCMA"/>
to
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=i...@20i,G722,PCMU,PCMA"/>
both inbound and outbound calls succeed.
Here is a SIP trace of a successful call after I apply the above changes:
Nov 23 18:16:51.844 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE
sip:+19725357...@ipipgw.ipdimensions.com:5060;user=phone;transport=UDP;maddr=168.75.202.246
SIP/2.0
v: SIP/2.0/UDP
65.243.172.245:5060;branch=z9hG4bK1d6dd953ef66469db06038ec3bd2ec49.6fb41dbb
Record-Route: <sip:65.243.172.245:5060;lr>
v: SIP/2.0/UDP
65.217.40.205:5060;branch=z9hG4bK74e3354e4bab8f6d8afec83d314c15b8.6f41c9d4;received=65.217.40.205
record-route: <sip:65.217.40.205;lr>
f:
<sip:+19729831...@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3da425-183eff65-3da425
t: <sip:+19725357...@65.217.40.205:5060;user=phone>
i: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457
CSeq: 1 INVITE
Max-Forwards: 16
k: 100rel, replaces
allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
v: SIP/2.0/UDP
DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da425-f0c931a9-52a4c353;received=199.173.101.208
m: <sip:199.173.101.208:5060;transport=UDP>
c: application/SDP
l: 210
P-Asserted-Identity: <sip:9729831...@65.217.40.205;user=phone>
Privacy: none
v=0
o=- 654094598 654094598 IN IP4 199.173.111.138
s=-
c=IN IP4 199.173.111.138
t=0 0
m=audio 31456 RTP/AVP 18 0 8 101
a=ptime:20
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 23 18:16:51.852 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
65.243.172.245:5060;branch=z9hG4bK1d6dd953ef66469db06038ec3bd2ec49.6fb41dbb,SIP/2.0/UDP
65.217.40.205:5060;branch=z9hG4bK74e3354e4bab8f6d8afec83d314c15b8.6f41c9d4;received=65.217.40.205,SIP/2.0/UDP
DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da425-f0c931a9-52a4c353;received=199.173.101.208
From:
<sip:+19729831...@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3da425-183eff65-3da425
To: <sip:+19725357...@65.217.40.205:5060;user=phone>
Date: Tue, 24 Nov 2009 00:16:51 GMT
Call-ID: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Nov 23 18:16:51.856 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:19725357...@168.75.202.212:5062 SIP/2.0
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AD9D0
From: <sip:19729831...@168.75.202.246>;tag=106FC130-1578
To: <sip:19725357...@168.75.202.212>
Date: Tue, 24 Nov 2009 00:16:51 GMT
Call-ID: 7fb1015e-d7c511de-a2ccd9df-3419a...@168.75.202.246
Supported: timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 2142226702-3620016606-2730940895-874095366
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1259021811
Contact: <sip:19729831...@168.75.202.246:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 15
P-Asserted-Identity: <sip:19729831...@168.75.202.246>
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 314
v=0
o=CiscoSystemsSIP-GW-UserAgent 9668 3852 IN IP4 168.75.202.246
s=SIP Call
c=IN IP4 199.173.111.138
t=0 0
m=audio 31456 RTP/AVP 18 0 8 101
c=IN IP4 199.173.111.138
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 23 18:16:51.856 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AD9D0
From: <sip:19729831...@168.75.202.246>;tag=106FC130-1578
To: <sip:19725357...@168.75.202.212>
Call-ID: 7fb1015e-d7c511de-a2ccd9df-3419a...@168.75.202.246
CSeq: 101 INVITE
Timestamp: 1259021811 0.000356
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M
Content-Length: 0
Nov 23 18:16:51.908 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AD9D0
From: <sip:19729831...@168.75.202.246>;tag=106FC130-1578
To: <sip:19725357...@168.75.202.212>;tag=mXNXUN859rKBa
Call-ID: 7fb1015e-d7c511de-a2ccd9df-3419a...@168.75.202.246
CSeq: 101 INVITE
Contact: <sip:19725357...@168.75.202.212:5062;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Content-Length: 0
P-Asserted-Identity: "19725357722" <sip:19725357...@168.75.202.212>
Nov 23 18:16:51.908 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
65.243.172.245:5060;branch=z9hG4bK1d6dd953ef66469db06038ec3bd2ec49.6fb41dbb,SIP/2.0/UDP
65.217.40.205:5060;branch=z9hG4bK74e3354e4bab8f6d8afec83d314c15b8.6f41c9d4;received=65.217.40.205,SIP/2.0/UDP
DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da425-f0c931a9-52a4c353;received=199.173.101.208
From:
<sip:+19729831...@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3da425-183eff65-3da425
To: <sip:+19725357...@65.217.40.205:5060;user=phone>;tag=106FC168-50D
Date: Tue, 24 Nov 2009 00:16:51 GMT
Call-ID: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:19725357...@168.75.202.246:5060>
Record-Route: <sip:65.243.172.245:5060;lr>,<sip:65.217.40.205;lr>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Nov 23 18:16:54.408 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AD9D0
From: <sip:19729831...@168.75.202.246>;tag=106FC130-1578
To: <sip:19725357...@168.75.202.212>;tag=mXNXUN859rKBa
Call-ID: 7fb1015e-d7c511de-a2ccd9df-3419a...@168.75.202.246
CSeq: 101 INVITE
Contact: <sip:19725357...@168.75.202.212:5062;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Min-SE: 1800
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 251
P-Asserted-Identity: "19725357722" <sip:19725357...@168.75.202.212>
v=0
o=FreeSWITCH 1259003870 1259003871 IN IP4 168.75.202.212
s=FreeSWITCH
c=IN IP4 168.75.202.212
t=0 0
m=audio 17544 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
Nov 23 18:16:54.412 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:19725357...@168.75.202.212:5062;transport=udp SIP/2.0
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AE109A
From: <sip:19729831...@168.75.202.246>;tag=106FC130-1578
To: <sip:19725357...@168.75.202.212>;tag=mXNXUN859rKBa
Date: Tue, 24 Nov 2009 00:16:51 GMT
Call-ID: 7fb1015e-d7c511de-a2ccd9df-3419a...@168.75.202.246
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 23 18:16:54.412 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
65.243.172.245:5060;branch=z9hG4bK1d6dd953ef66469db06038ec3bd2ec49.6fb41dbb,SIP/2.0/UDP
65.217.40.205:5060;branch=z9hG4bK74e3354e4bab8f6d8afec83d314c15b8.6f41c9d4;received=65.217.40.205,SIP/2.0/UDP
DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da425-f0c931a9-52a4c353;received=199.173.101.208
From:
<sip:+19729831...@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3da425-183eff65-3da425
To: <sip:+19725357...@65.217.40.205:5060;user=phone>;tag=106FC168-50D
Date: Tue, 24 Nov 2009 00:16:51 GMT
Call-ID: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:19725357...@168.75.202.246:5060>
Record-Route: <sip:65.243.172.245:5060;lr>,<sip:65.217.40.205;lr>
Supported: replaces
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 253
v=0
o=CiscoSystemsSIP-GW-UserAgent 7353 3710 IN IP4 168.75.202.246
s=SIP Call
c=IN IP4 168.75.202.212
t=0 0
m=audio 17544 RTP/AVP 0 101
c=IN IP4 168.75.202.212
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Nov 23 18:16:54.492 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:19725357...@168.75.202.246:5060 SIP/2.0
v: SIP/2.0/UDP
65.243.172.245:5060;branch=z9hG4bKb17fac77c446113b9154e16639d30287.6be1820d
v: SIP/2.0/UDP
65.217.40.205:5060;branch=z9hG4bK99af095cadf31f4291c6a809ef6a6e03.7c44d9fc;received=65.217.40.205
f:
<sip:+19729831...@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3da425-183eff65-3da425
t: <sip:+19725357...@65.217.40.205:5060;user=phone>;tag=106FC168-50D
i: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457
CSeq: 1 ACK
user-agent: CS2000_NGSS/9.0
Max-Forwards: 68
k: 100rel,replaces
allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
v: SIP/2.0/UDP
DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da428-f0c93cc6-61239372;received=199.173.101.208
m: <sip:199.173.101.208:5060;transport=UDP>
l: 0
Nov 23 18:17:00.636 CST: %FAN-3-FAN_FAILED: Fan 1 had a rotation error reported.
Nov 23 18:17:06.748 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:19729831...@168.75.202.246:5060 SIP/2.0
Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKmNaaUQK5vrpXQ
Max-Forwards: 70
From: <sip:19725357...@168.75.202.212>;tag=mXNXUN859rKBa
To: <sip:19729831...@168.75.202.246>;tag=106FC130-1578
Call-ID: 7fb1015e-d7c511de-a2ccd9df-3419a...@168.75.202.246
CSeq: 123392377 BYE
Contact: <sip:19725357...@168.75.202.212:5062;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
Nov 23 18:17:06.748 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKmNaaUQK5vrpXQ
From: <sip:19725357...@168.75.202.212>;tag=mXNXUN859rKBa
To: <sip:19729831...@168.75.202.246>;tag=106FC130-1578
Date: Tue, 24 Nov 2009 00:17:06 GMT
Call-ID: 7fb1015e-d7c511de-a2ccd9df-3419a...@168.75.202.246
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 123392377 BYE
Reason: Q.850;cause=16
Content-Length: 0
Nov 23 18:17:06.752 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:199.173.101.208:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AF4FA
From: <sip:+19725357...@65.217.40.205:5060;user=phone>;tag=106FC168-50D
To:
<sip:+19729831...@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3da425-183eff65-3da425
Date: Tue, 24 Nov 2009 00:16:54 GMT
Call-ID: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Route: <sip:65.243.172.245:5060;lr>,<sip:65.217.40.205;lr>
Timestamp: 1259021826
CSeq: 101 BYE
Reason: Q.850;cause=16
Content-Length: 0
Nov 23 18:17:06.824 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
f: <sip:+19725357...@65.217.40.205:5060;user=phone>;tag=106FC168-50D
t:
<sip:+19729831...@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3da425-183eff65-3da425
i: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457
CSeq: 101 BYE
server: CS2000_NGSS/9.0
allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
v: SIP/2.0/UDP 168.75.202.246:5060;received=168.75.202.246;branch=z9hG4bK65AF4FA
l: 0
I compared the SIP messaging from the failed call to the SIP messaging from the
good call. Both calls are inbound calls.
Here is the session description for the failed inbound call:
v=0
o=- 211627 211627 IN IP4 192.168.1.4
s=-
c=IN IP4 173.57.44.212
t=0 0
m=audio 0 RTP/AVP 96 101
a=rtpmap:96 G729a/8000
a=fmtp:96 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Here is the session description for the good outbound call:
v=0
o=FreeSWITCH 1259003870 1259003871 IN IP4 168.75.202.212
s=FreeSWITCH
c=IN IP4 168.75.202.212
t=0 0
m=audio 17544 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
Here are the differences between the session descriptors of the failed call and
the good call:
- The c= line has the correct IP address for the failed call, which was using
media bypass
- The c= line has the correct IP address for the good call, because the media
is being processed
by FreeSWITCH in the good call
- The m= line does not have the correct RTP port in the failed call
- The m= line has the correct RTP port in the good call
I noticed that the SDP media descriptor is incorrect in the failed call. Has
this problem been fixed? I am running revision 15586 from the FreeSWITCH SVN
trunk.
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