No voice Behind NAT
Moderator: Brekeke Support Team
No voice Behind NAT
1. Brekeke Product Name and Version: v3.1.7.8
2. Java version:
3. OS type and the version: Windows 2003
4. UA (phone), gateway or other hardware/software involved:
5. Your problem:
i have two IP client running on android mobile.both are registered successfully using public IP address.and both clients are behind NAT in WIFI networks.and call also successfully established.but no voice in both endpoints.both clients sends RTP to sip server.but sip server not forwarding that to both the endpoints. kindly help me to solve this issue.
Regards
Bensly
2. Java version:
3. OS type and the version: Windows 2003
4. UA (phone), gateway or other hardware/software involved:
5. Your problem:
i have two IP client running on android mobile.both are registered successfully using public IP address.and both clients are behind NAT in WIFI networks.and call also successfully established.but no voice in both endpoints.both clients sends RTP to sip server.but sip server not forwarding that to both the endpoints. kindly help me to solve this issue.
Regards
Bensly
How about the [Port mapping] setting in the [RTP] page?
It should be "source port" in your case.
If you try another SIP client such as Linphone..
Does the same issue happen?
Also, let you check your firewall settings.
Make sure that UDP ports 10000-29999 should be opened at Window Firewall.
http://wiki.brekeke.com/wiki/Using-Brek ... a-firewall
It should be "source port" in your case.
If you try another SIP client such as Linphone..
Does the same issue happen?
Also, let you check your firewall settings.
Make sure that UDP ports 10000-29999 should be opened at Window Firewall.
http://wiki.brekeke.com/wiki/Using-Brek ... a-firewall
Hi,
[Port Mapping] settings selected as source port only.and RTP relay mode is on.still same problem.client using local IP address in the SDP,but sip server not replacing the public IP address. i have tested with asterisk server.in that case all the call audio flows both side and works well.i don't know why brekeke sip server not replacing the public IP address in the SDP.
Regards
Bensly
[Port Mapping] settings selected as source port only.and RTP relay mode is on.still same problem.client using local IP address in the SDP,but sip server not replacing the public IP address. i have tested with asterisk server.in that case all the call audio flows both side and works well.i don't know why brekeke sip server not replacing the public IP address in the SDP.
Regards
Bensly
> but sip server not replacing the public IP address.
It seems that the SIP Server doesn't know its global IP address, or RTP-relay is not enabled yet.
Is the Brekeke SIP Server behind NAT?
If yes, you need to define its global IP address at the [Interface address 1] in the [Configuration]->[System] page.
And you need to restart the SIP Server after you modified configuration.
It seems that the SIP Server doesn't know its global IP address, or RTP-relay is not enabled yet.
Is the Brekeke SIP Server behind NAT?
If yes, you need to define its global IP address at the [Interface address 1] in the [Configuration]->[System] page.
And you need to restart the SIP Server after you modified configuration.
Sip server behind NAT,and i have configured local and public IP in interface 1 and 2. and RTP replay is ON , RTP Relay ( UA) is auto.port mapping is source port, and send UA remote address is yes.
and after configured this settings i restarted the sip server also.it not replacing the public IP in SDP,it replaced local IP of the sip server only. please find the Invite request.
INVITE sip:600@59.90.246.89 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.199:57099;branch=z9hG4bK1014799920;rport
From: <sip:4040@59.90.246.89>;tag=819769433
To: <sip:600@59.90.246.89>
Contact: <sip:4040@192.168.1.199:57099;transport=udp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: bff97531-cf8e-1cac-98aa-e3fc10163731
CSeq: 2117188504 INVITE
Content-Type: application/sdp
Content-Length: 449
Max-Forwards: 70
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.540.831 (doubango r831 - Micromax A110)
P-Preferred-Identity: <sip:4040@59.90.246.89>
v=0
o=doubango 1983 678901 IN IP4 192.168.1.199
s=-
c=IN IP4 192.168.1.199
t=0 0
a=tcap:1 RTP/AVPF
m=audio 16298 RTP/AVP 8 0 101
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=pcfg:1 t=1
a=sendrecv
a=ssrc:3332805306 cname:ldjWoB60jbyQlR6e
a=ssrc:3332805306 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:3332805306 label:Doubango.Audio
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.199:57099;branch=z9hG4bK1014799920;rport=27608;received=59.90.246.89
From: <sip:4040@59.90.246.89>;tag=819769433
To: <sip:600@59.90.246.89>
Call-ID: bff97531-cf8e-1cac-98aa-e3fc10163731
CSeq: 2117188504 INVITE
Server: Brekeke SIP Server rev.348.2 Evaluation
Content-Length: 0
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.199:57099;branch=z9hG4bK1014799920;rport=27608;received=59.90.246.89
From: <sip:4040@59.90.246.89>;tag=819769433
To: <sip:600@59.90.246.89>;tag=ba699354fs
Contact: <sip:600@192.168.1.237:5061>
Call-ID: bff97531-cf8e-1cac-98aa-e3fc10163731
CSeq: 2117188504 INVITE
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
Content-Length: 0
....SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.199:57099;branch=z9hG4bK1014799920;rport=27608;received=59.90.246.89
From: <sip:4040@59.90.246.89>;tag=819769433
To: <sip:600@59.90.246.89>;tag=ba699354fs
Contact: <sip:600@192.168.1.237:5061>
Call-ID: bff97531-cf8e-1cac-98aa-e3fc10163731
CSeq: 2117188504 INVITE
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 431
v=0
o=doubango 1983 678901 IN IP4 192.168.1.237
s=-
c=IN IP4 192.168.1.237
t=0 0
m=audio 10020 RTP/AVPF 8 0 101
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=acfg:1 t=1
a=sendrecv
a=ssrc:4107553691 cname:ldjWoB60jbyQlR6e
a=ssrc:4107553691 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:4107553691 label:Doubango.Audio
and after configured this settings i restarted the sip server also.it not replacing the public IP in SDP,it replaced local IP of the sip server only. please find the Invite request.
INVITE sip:600@59.90.246.89 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.199:57099;branch=z9hG4bK1014799920;rport
From: <sip:4040@59.90.246.89>;tag=819769433
To: <sip:600@59.90.246.89>
Contact: <sip:4040@192.168.1.199:57099;transport=udp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: bff97531-cf8e-1cac-98aa-e3fc10163731
CSeq: 2117188504 INVITE
Content-Type: application/sdp
Content-Length: 449
Max-Forwards: 70
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.540.831 (doubango r831 - Micromax A110)
P-Preferred-Identity: <sip:4040@59.90.246.89>
v=0
o=doubango 1983 678901 IN IP4 192.168.1.199
s=-
c=IN IP4 192.168.1.199
t=0 0
a=tcap:1 RTP/AVPF
m=audio 16298 RTP/AVP 8 0 101
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=pcfg:1 t=1
a=sendrecv
a=ssrc:3332805306 cname:ldjWoB60jbyQlR6e
a=ssrc:3332805306 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:3332805306 label:Doubango.Audio
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.199:57099;branch=z9hG4bK1014799920;rport=27608;received=59.90.246.89
From: <sip:4040@59.90.246.89>;tag=819769433
To: <sip:600@59.90.246.89>
Call-ID: bff97531-cf8e-1cac-98aa-e3fc10163731
CSeq: 2117188504 INVITE
Server: Brekeke SIP Server rev.348.2 Evaluation
Content-Length: 0
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.199:57099;branch=z9hG4bK1014799920;rport=27608;received=59.90.246.89
From: <sip:4040@59.90.246.89>;tag=819769433
To: <sip:600@59.90.246.89>;tag=ba699354fs
Contact: <sip:600@192.168.1.237:5061>
Call-ID: bff97531-cf8e-1cac-98aa-e3fc10163731
CSeq: 2117188504 INVITE
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
Content-Length: 0
....SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.199:57099;branch=z9hG4bK1014799920;rport=27608;received=59.90.246.89
From: <sip:4040@59.90.246.89>;tag=819769433
To: <sip:600@59.90.246.89>;tag=ba699354fs
Contact: <sip:600@192.168.1.237:5061>
Call-ID: bff97531-cf8e-1cac-98aa-e3fc10163731
CSeq: 2117188504 INVITE
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 431
v=0
o=doubango 1983 678901 IN IP4 192.168.1.237
s=-
c=IN IP4 192.168.1.237
t=0 0
m=audio 10020 RTP/AVPF 8 0 101
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=acfg:1 t=1
a=sendrecv
a=ssrc:4107553691 cname:ldjWoB60jbyQlR6e
a=ssrc:4107553691 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:4107553691 label:Doubango.Audio
The IP address as you said is correct. and in Interface also listed my local and public IP. for more information i have enabled the sip server logs. please find the log below to get identify the problem.
================================================================================
Brekeke SIP Server 3.1/348.2
Copyright (C) 2002-2013 Brekeke Software, Inc. All rights reserved.
================================================================================
sv: open logging-file: 'C:/Program Files/Brekeke/sip/webapps/sip/WEB-INF/work/sv/log/2013/04/sv.20130410.1.log'
sv: logging-plugin: com.brekeke.common.Logging
sv: 'your-sip-sv' at 'your-place' is starting...
sv: os=Windows 2003 (x86:5.2) java=1.6.0_17 (Sun Microsystems Inc.)
sv: total.mem=5177344 free.mem=4671840 cpu=2
svlistener: start at 04/10/13 18:26:43.203
tcp-listener: start
tcp-listener: listen-port=5061
svlistener: open session-log 'C:/Program Files/Brekeke/sip/webapps/sip/WEB-INF/work/sv/log/2013/04/session.20130410.log'.
svlistener: open dial-plan 'C:/Program Files/Brekeke/sip/webapps/sip/WEB-INF/work/sv/etc/dialplan.tbl'.
svlistener: hostname=bensly-desktop listen-port=5061
svlistener: IPv4: preferIPv6Addresses=false preferIPv4Stack=true
svlistener: interface={ 192.168.1.237, 59.90.246.89 }
session.12: sipex.11: start: from=<sip:668@59.90.246.89> to=<sip:700@192.168.1.237:5061>
session.12: information:
starttime = 04/10/13 18:27:05.109
spiral-hop = 1
plugin = com.brekeke.net.sip.sv.session.plugins.InviteSession
request = INVITE sip:700@59.90.246.89 SIP/2.0
rulename = registered=sip:700(sip:700@192.168.1.199:53611/UDP)
org:From: = sip:668@59.90.246.89
new:From: = sip:668@59.90.246.89
org:To: = sip:700@59.90.246.89
new:To: = sip:700@192.168.1.237:5061
src:addr/port = 59.90.246.89:14334 (UDP global-addr if)
src:interface = 192.168.1.237:5061 (UDP local-addr)
dst:addr/port = 59.90.246.89:29494 (UDP global-addr if)
dst:interface = 192.168.1.237:5061 (UDP local-addr)
mode:B2BUA = off
mode:RTPrelay = auto
mode:Auth = auto
mode:NAT = on { Src-Far-End-NAT Dst-Far-End-NAT }
session.12: phase=0: Initializing
session.12: System Used Memory = 2787
session.12: receive: from=UAC:59.90.246.89:14334(UDP) at 04/10/13 18:27:05.109
==============================================
INVITE sip:700@59.90.246.89 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.226:36306;branch=z9hG4bK1751344313;rport=14334;received=59.90.246.89
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@59.90.246.89>
Contact: <sip:668@192.168.1.226:36306;transport=udp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 INVITE
Max-Forwards: 70
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.532.758 (doubango r758 - Micromax A110)
P-Preferred-Identity: <sip:668@59.90.246.89>
Supported: 100rel
P-Behind-NAT: Yes
Content-Type: application/sdp
Content-Length: 429
v=0
o=doubango 1983 678901 IN IP4 192.168.1.226
s=-
c=IN IP4 192.168.1.226
t=0 0
m=audio 27142 RTP/AVP 9 101
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=tcap:1 RTP/AVPF
a=pcfg:1 t=1
a=sendrecv
a=rtcp-mux
a=ssrc:2673671146 cname:ldjWoB60jbyQlR6e
a=ssrc:2673671146 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2673671146 label:Doubango
==============================================
session.12: content-type=application/sdp plugin=com.brekeke.net.content.application.Sdp
session.12: pkt=1 dp=1 st=0 sip:668@59.90.246.89(59.90.246.89:14334) --> sip:700@192.168.1.237:5061(59.90.246.89:29494)
send="INVITE sip:700@192.168.1.199:53611;transport=udp SIP/2.0"
session.12: phase=1: Inviting
session.12: processtime=4516
session.12: send: to=UAS:59.90.246.89:29494(UDP) at 04/10/13 18:27:05.109
==============================================
INVITE sip:700@192.168.1.199:53611;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bK235c46dc205ac-30-2d73c1a
Via: SIP/2.0/UDP 192.168.1.226:36306;branch=z9hG4bK1751344313;rport=14334;received=59.90.246.89
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>
Contact: <sip:668@192.168.1.237:5061>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 INVITE
Max-Forwards: 69
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.532.758 (doubango r758 - Micromax A110)
P-Preferred-Identity: <sip:668@59.90.246.89>
Supported: 100rel
P-Behind-NAT: Yes
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Type: application/sdp
Content-Length: 429
v=0
o=doubango 1983 678901 IN IP4 192.168.1.237
s=-
c=IN IP4 192.168.1.237
t=0 0
m=audio 10000 RTP/AVP 9 101
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=tcap:1 RTP/AVPF
a=pcfg:1 t=1
a=sendrecv
a=rtcp-mux
a=ssrc:2673671146 cname:ldjWoB60jbyQlR6e
a=ssrc:2673671146 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2673671146 label:Doubango
==============================================
session.12: receive: from=UAS:192.168.1.199:53611(UDP) at 04/10/13 18:27:05.156
==============================================
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bK235c46dc205ac-30-2d73c1a
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 INVITE
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=14334;received=59.90.246.89;branch=z9hG4bK1751344313
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Length: 0
==============================================
session.12: pkt=2 dp=2 st=0 sip:700@192.168.1.237:5061(192.168.1.199:53611) --> sip:668@59.90.246.89(59.90.246.89:14334)
send="SIP/2.0 100 Trying (sent from the Transaction Layer)"
session.12: phase=2: Provisioning
session.12: total_spiral_hops=1
session.12: receive: from=UAS:192.168.1.199:53611(UDP) at 04/10/13 18:27:05.171
==============================================
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bK235c46dc205ac-30-2d73c1a
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>;tag=1496733710
Contact: <sip:700@192.168.1.199:53611;transport=udp>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 INVITE
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=14334;received=59.90.246.89;branch=z9hG4bK1751344313
Record-Route: <sip:192.168.1.237:5061;lr>
Require: 100rel
RSeq: 731372144
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
Content-Length: 0
==============================================
session.12: pkt=3 dp=2 st=0 sip:700@192.168.1.237:5061(192.168.1.199:53611) --> sip:668@59.90.246.89(59.90.246.89:14334)
send="SIP/2.0 180 Ringing"
session.12: phase=3: Ringing
session.12: processtime=0
session.12: send: to=UAC:59.90.246.89:14334(UDP) at 04/10/13 18:27:05.171
==============================================
SIP/2.0 180 Ringing
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@59.90.246.89>;tag=1496733710
Contact: <sip:700@192.168.1.237:5061>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 INVITE
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=14334;received=59.90.246.89;branch=z9hG4bK1751344313
Record-Route: <sip:192.168.1.237:5061;lr>
Require: 100rel
RSeq: 731372144
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
Content-Length: 0
==============================================
session.12: receive: from=UAC:192.168.1.226:36306(UDP) at 04/10/13 18:27:05.203
==============================================
PRACK sip:700@192.168.1.237:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.226:36306;branch=z9hG4bK193301599;rport=36306
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@59.90.246.89>;tag=1496733710
Contact: <sip:668@192.168.1.226:36306;transport=udp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223845 PRACK
Max-Forwards: 70
Route: <sip:192.168.1.237:5061;lr>
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.532.758 (doubango r758 - Micromax A110)
RAck: 731372144 223844 INVITE
Content-Length: 0
==============================================
session.12: pkt=4 dp=1 st=0 sip:668@59.90.246.89(192.168.1.226:36306) --> sip:700@192.168.1.237:5061(192.168.1.199:53611)
send="PRACK sip:700@192.168.1.199:53611;transport=udp SIP/2.0"
session.12: processtime=0
session.12: send: to=UAS:192.168.1.199:53611(UDP) at 04/10/13 18:27:05.203
==============================================
PRACK sip:700@192.168.1.199:53611;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bK327666c6205ac-30-2d73c1a
Via: SIP/2.0/UDP 192.168.1.226:36306;branch=z9hG4bK193301599;rport=36306
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>;tag=1496733710
Contact: <sip:668@192.168.1.237:5061>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223845 PRACK
Max-Forwards: 69
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.532.758 (doubango r758 - Micromax A110)
RAck: 731372144 223844 INVITE
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Length: 0
==============================================
session.12: receive: from=UAS:192.168.1.199:53611(UDP) at 04/10/13 18:27:05.328
==============================================
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bK327666c6205ac-30-2d73c1a
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>;tag=1496733710
Contact: <sip:700@192.168.1.199:53611;transport=udp>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223845 PRACK
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=36306;branch=z9hG4bK193301599
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Length: 0
==============================================
session.12: pkt=5 dp=2 st=0 sip:700@192.168.1.237:5061(192.168.1.199:53611) --> sip:668@59.90.246.89(192.168.1.226:36306)
send="SIP/2.0 200 OK"
session.12: processtime=0
session.12: send: to=UAC:192.168.1.226:36306(UDP) at 04/10/13 18:27:05.328
==============================================
SIP/2.0 200 OK
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@59.90.246.89>;tag=1496733710
Contact: <sip:700@192.168.1.237:5061>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223845 PRACK
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=36306;branch=z9hG4bK193301599
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Length: 0
==============================================
session.12: receive: from=UAS:192.168.1.199:53611(UDP) at 04/10/13 18:27:08.531
==============================================
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bK235c46dc205ac-30-2d73c1a
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>;tag=1496733710
Contact: <sip:700@192.168.1.199:53611;transport=udp>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 INVITE
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=14334;received=59.90.246.89;branch=z9hG4bK1751344313
Record-Route: <sip:192.168.1.237:5061;lr>
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 417
v=0
o=doubango 1983 678901 IN IP4 192.168.1.199
s=-
c=IN IP4 192.168.1.199
t=0 0
m=audio 65534 RTP/AVPF 9 101
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=acfg:1 t=1
a=sendrecv
a=rtcp-mux
a=ssrc:2174826863 cname:ldjWoB60jbyQlR6e
a=ssrc:2174826863 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2174826863 label:Doubango.Audio
==============================================
session.12: content-type=application/sdp plugin=com.brekeke.net.content.application.Sdp
session.12: pkt=6 dp=2 st=0 sip:700@192.168.1.237:5061(192.168.1.199:53611) --> sip:668@59.90.246.89(59.90.246.89:14334)
send="SIP/2.0 200 OK"
session.12: phase=4: Accepted
session.12: processtime=0
session.12: send: to=UAC:59.90.246.89:14334(UDP) at 04/10/13 18:27:08.531
==============================================
SIP/2.0 200 OK
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@59.90.246.89>;tag=1496733710
Contact: <sip:700@192.168.1.237:5061>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 INVITE
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=14334;received=59.90.246.89;branch=z9hG4bK1751344313
Record-Route: <sip:192.168.1.237:5061;lr>
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 417
v=0
o=doubango 1983 678901 IN IP4 192.168.1.237
s=-
c=IN IP4 192.168.1.237
t=0 0
m=audio 10002 RTP/AVPF 9 101
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=acfg:1 t=1
a=sendrecv
a=rtcp-mux
a=ssrc:2174826863 cname:ldjWoB60jbyQlR6e
a=ssrc:2174826863 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2174826863 label:Doubango.Audio
==============================================
session.12: receive: from=UAC:192.168.1.226:36306(UDP) at 04/10/13 18:27:08.625
==============================================
ACK sip:700@192.168.1.237:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.226:36306;branch=z9hG4bK486036623;rport=36306
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@59.90.246.89>;tag=1496733710
Contact: <sip:668@192.168.1.226:36306;transport=udp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 ACK
Max-Forwards: 70
Route: <sip:192.168.1.237:5061;lr>
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.532.758 (doubango r758 - Micromax A110)
Content-Length: 0
==============================================
session.12: pkt=7 dp=1 st=0 sip:668@59.90.246.89(192.168.1.226:36306) --> sip:700@192.168.1.237:5061(192.168.1.199:53611)
send="ACK sip:700@192.168.1.199:53611;transport=udp SIP/2.0"
session.12: phase=5: Talking
session.12: processtime=0
session.12: send: to=UAS:192.168.1.199:53611(UDP) at 04/10/13 18:27:08.625
==============================================
ACK sip:700@192.168.1.199:53611;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bK21233e0f205ac-30-2d73c1a
Via: SIP/2.0/UDP 192.168.1.226:36306;branch=z9hG4bK486036623;rport=36306
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>;tag=1496733710
Contact: <sip:668@192.168.1.237:5061>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 ACK
Max-Forwards: 69
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.532.758 (doubango r758 - Micromax A110)
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Length: 0
==============================================
session.12: receive: from=UAC:192.168.1.226:36306(UDP) at 04/10/13 18:27:19.296
==============================================
BYE sip:700@192.168.1.237:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.226:36306;branch=z9hG4bK309144096;rport=36306
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@59.90.246.89>;tag=1496733710
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223846 BYE
Max-Forwards: 70
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Route: <sip:192.168.1.237:5061;lr>
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.532.758 (doubango r758 - Micromax A110)
P-Preferred-Identity: <sip:668@59.90.246.89>
Content-Length: 0
==============================================
session.12: pkt=8 dp=1 st=0 sip:668@59.90.246.89(192.168.1.226:36306) --> sip:700@192.168.1.237:5061(192.168.1.199:53611)
send="BYE sip:700@192.168.1.199:53611;transport=udp SIP/2.0"
session.12: phase=6: Closing
session.12: processtime=0
session.12: send: to=UAS:192.168.1.199:53611(UDP) at 04/10/13 18:27:19.296
==============================================
BYE sip:700@192.168.1.199:53611;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bKee4a0b93205ac-30-2d73c1a
Via: SIP/2.0/UDP 192.168.1.226:36306;branch=z9hG4bK309144096;rport=36306
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>;tag=1496733710
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223846 BYE
Max-Forwards: 69
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.532.758 (doubango r758 - Micromax A110)
P-Preferred-Identity: <sip:668@59.90.246.89>
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Length: 0
==============================================
session.12: receive: from=UAS:192.168.1.199:53611(UDP) at 04/10/13 18:27:19.312
==============================================
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bKee4a0b93205ac-30-2d73c1a
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>;tag=1496733710
Contact: <sip:700@192.168.1.199:53611;transport=udp>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223846 BYE
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=36306;branch=z9hG4bK309144096
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Length: 0
==============================================
session.12: pkt=9 dp=2 st=0 sip:700@192.168.1.237:5061(192.168.1.199:53611) --> sip:668@59.90.246.89(192.168.1.226:36306)
send="SIP/2.0 200 OK"
session.12: processtime=0
session.12: send: to=UAC:192.168.1.226:36306(UDP) at 04/10/13 18:27:19.312
==============================================
SIP/2.0 200 OK
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@59.90.246.89>;tag=1496733710
Contact: <sip:700@192.168.1.237:5061>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223846 BYE
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=36306;branch=z9hG4bK309144096
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Length: 0
==============================================
session.12: sipex.11: close: result=Success length=00:00:10 total-pkt=9 at 04/10/13 18:27:19.312
================================================================================
Brekeke SIP Server 3.1/348.2
Copyright (C) 2002-2013 Brekeke Software, Inc. All rights reserved.
================================================================================
sv: open logging-file: 'C:/Program Files/Brekeke/sip/webapps/sip/WEB-INF/work/sv/log/2013/04/sv.20130410.1.log'
sv: logging-plugin: com.brekeke.common.Logging
sv: 'your-sip-sv' at 'your-place' is starting...
sv: os=Windows 2003 (x86:5.2) java=1.6.0_17 (Sun Microsystems Inc.)
sv: total.mem=5177344 free.mem=4671840 cpu=2
svlistener: start at 04/10/13 18:26:43.203
tcp-listener: start
tcp-listener: listen-port=5061
svlistener: open session-log 'C:/Program Files/Brekeke/sip/webapps/sip/WEB-INF/work/sv/log/2013/04/session.20130410.log'.
svlistener: open dial-plan 'C:/Program Files/Brekeke/sip/webapps/sip/WEB-INF/work/sv/etc/dialplan.tbl'.
svlistener: hostname=bensly-desktop listen-port=5061
svlistener: IPv4: preferIPv6Addresses=false preferIPv4Stack=true
svlistener: interface={ 192.168.1.237, 59.90.246.89 }
session.12: sipex.11: start: from=<sip:668@59.90.246.89> to=<sip:700@192.168.1.237:5061>
session.12: information:
starttime = 04/10/13 18:27:05.109
spiral-hop = 1
plugin = com.brekeke.net.sip.sv.session.plugins.InviteSession
request = INVITE sip:700@59.90.246.89 SIP/2.0
rulename = registered=sip:700(sip:700@192.168.1.199:53611/UDP)
org:From: = sip:668@59.90.246.89
new:From: = sip:668@59.90.246.89
org:To: = sip:700@59.90.246.89
new:To: = sip:700@192.168.1.237:5061
src:addr/port = 59.90.246.89:14334 (UDP global-addr if)
src:interface = 192.168.1.237:5061 (UDP local-addr)
dst:addr/port = 59.90.246.89:29494 (UDP global-addr if)
dst:interface = 192.168.1.237:5061 (UDP local-addr)
mode:B2BUA = off
mode:RTPrelay = auto
mode:Auth = auto
mode:NAT = on { Src-Far-End-NAT Dst-Far-End-NAT }
session.12: phase=0: Initializing
session.12: System Used Memory = 2787
session.12: receive: from=UAC:59.90.246.89:14334(UDP) at 04/10/13 18:27:05.109
==============================================
INVITE sip:700@59.90.246.89 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.226:36306;branch=z9hG4bK1751344313;rport=14334;received=59.90.246.89
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@59.90.246.89>
Contact: <sip:668@192.168.1.226:36306;transport=udp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 INVITE
Max-Forwards: 70
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.532.758 (doubango r758 - Micromax A110)
P-Preferred-Identity: <sip:668@59.90.246.89>
Supported: 100rel
P-Behind-NAT: Yes
Content-Type: application/sdp
Content-Length: 429
v=0
o=doubango 1983 678901 IN IP4 192.168.1.226
s=-
c=IN IP4 192.168.1.226
t=0 0
m=audio 27142 RTP/AVP 9 101
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=tcap:1 RTP/AVPF
a=pcfg:1 t=1
a=sendrecv
a=rtcp-mux
a=ssrc:2673671146 cname:ldjWoB60jbyQlR6e
a=ssrc:2673671146 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2673671146 label:Doubango
==============================================
session.12: content-type=application/sdp plugin=com.brekeke.net.content.application.Sdp
session.12: pkt=1 dp=1 st=0 sip:668@59.90.246.89(59.90.246.89:14334) --> sip:700@192.168.1.237:5061(59.90.246.89:29494)
send="INVITE sip:700@192.168.1.199:53611;transport=udp SIP/2.0"
session.12: phase=1: Inviting
session.12: processtime=4516
session.12: send: to=UAS:59.90.246.89:29494(UDP) at 04/10/13 18:27:05.109
==============================================
INVITE sip:700@192.168.1.199:53611;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bK235c46dc205ac-30-2d73c1a
Via: SIP/2.0/UDP 192.168.1.226:36306;branch=z9hG4bK1751344313;rport=14334;received=59.90.246.89
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>
Contact: <sip:668@192.168.1.237:5061>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 INVITE
Max-Forwards: 69
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.532.758 (doubango r758 - Micromax A110)
P-Preferred-Identity: <sip:668@59.90.246.89>
Supported: 100rel
P-Behind-NAT: Yes
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Type: application/sdp
Content-Length: 429
v=0
o=doubango 1983 678901 IN IP4 192.168.1.237
s=-
c=IN IP4 192.168.1.237
t=0 0
m=audio 10000 RTP/AVP 9 101
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=tcap:1 RTP/AVPF
a=pcfg:1 t=1
a=sendrecv
a=rtcp-mux
a=ssrc:2673671146 cname:ldjWoB60jbyQlR6e
a=ssrc:2673671146 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2673671146 label:Doubango
==============================================
session.12: receive: from=UAS:192.168.1.199:53611(UDP) at 04/10/13 18:27:05.156
==============================================
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bK235c46dc205ac-30-2d73c1a
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 INVITE
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=14334;received=59.90.246.89;branch=z9hG4bK1751344313
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Length: 0
==============================================
session.12: pkt=2 dp=2 st=0 sip:700@192.168.1.237:5061(192.168.1.199:53611) --> sip:668@59.90.246.89(59.90.246.89:14334)
send="SIP/2.0 100 Trying (sent from the Transaction Layer)"
session.12: phase=2: Provisioning
session.12: total_spiral_hops=1
session.12: receive: from=UAS:192.168.1.199:53611(UDP) at 04/10/13 18:27:05.171
==============================================
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bK235c46dc205ac-30-2d73c1a
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>;tag=1496733710
Contact: <sip:700@192.168.1.199:53611;transport=udp>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 INVITE
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=14334;received=59.90.246.89;branch=z9hG4bK1751344313
Record-Route: <sip:192.168.1.237:5061;lr>
Require: 100rel
RSeq: 731372144
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
Content-Length: 0
==============================================
session.12: pkt=3 dp=2 st=0 sip:700@192.168.1.237:5061(192.168.1.199:53611) --> sip:668@59.90.246.89(59.90.246.89:14334)
send="SIP/2.0 180 Ringing"
session.12: phase=3: Ringing
session.12: processtime=0
session.12: send: to=UAC:59.90.246.89:14334(UDP) at 04/10/13 18:27:05.171
==============================================
SIP/2.0 180 Ringing
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@59.90.246.89>;tag=1496733710
Contact: <sip:700@192.168.1.237:5061>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 INVITE
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=14334;received=59.90.246.89;branch=z9hG4bK1751344313
Record-Route: <sip:192.168.1.237:5061;lr>
Require: 100rel
RSeq: 731372144
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
Content-Length: 0
==============================================
session.12: receive: from=UAC:192.168.1.226:36306(UDP) at 04/10/13 18:27:05.203
==============================================
PRACK sip:700@192.168.1.237:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.226:36306;branch=z9hG4bK193301599;rport=36306
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@59.90.246.89>;tag=1496733710
Contact: <sip:668@192.168.1.226:36306;transport=udp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223845 PRACK
Max-Forwards: 70
Route: <sip:192.168.1.237:5061;lr>
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.532.758 (doubango r758 - Micromax A110)
RAck: 731372144 223844 INVITE
Content-Length: 0
==============================================
session.12: pkt=4 dp=1 st=0 sip:668@59.90.246.89(192.168.1.226:36306) --> sip:700@192.168.1.237:5061(192.168.1.199:53611)
send="PRACK sip:700@192.168.1.199:53611;transport=udp SIP/2.0"
session.12: processtime=0
session.12: send: to=UAS:192.168.1.199:53611(UDP) at 04/10/13 18:27:05.203
==============================================
PRACK sip:700@192.168.1.199:53611;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bK327666c6205ac-30-2d73c1a
Via: SIP/2.0/UDP 192.168.1.226:36306;branch=z9hG4bK193301599;rport=36306
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>;tag=1496733710
Contact: <sip:668@192.168.1.237:5061>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223845 PRACK
Max-Forwards: 69
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.532.758 (doubango r758 - Micromax A110)
RAck: 731372144 223844 INVITE
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Length: 0
==============================================
session.12: receive: from=UAS:192.168.1.199:53611(UDP) at 04/10/13 18:27:05.328
==============================================
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bK327666c6205ac-30-2d73c1a
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>;tag=1496733710
Contact: <sip:700@192.168.1.199:53611;transport=udp>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223845 PRACK
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=36306;branch=z9hG4bK193301599
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Length: 0
==============================================
session.12: pkt=5 dp=2 st=0 sip:700@192.168.1.237:5061(192.168.1.199:53611) --> sip:668@59.90.246.89(192.168.1.226:36306)
send="SIP/2.0 200 OK"
session.12: processtime=0
session.12: send: to=UAC:192.168.1.226:36306(UDP) at 04/10/13 18:27:05.328
==============================================
SIP/2.0 200 OK
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@59.90.246.89>;tag=1496733710
Contact: <sip:700@192.168.1.237:5061>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223845 PRACK
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=36306;branch=z9hG4bK193301599
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Length: 0
==============================================
session.12: receive: from=UAS:192.168.1.199:53611(UDP) at 04/10/13 18:27:08.531
==============================================
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bK235c46dc205ac-30-2d73c1a
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>;tag=1496733710
Contact: <sip:700@192.168.1.199:53611;transport=udp>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 INVITE
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=14334;received=59.90.246.89;branch=z9hG4bK1751344313
Record-Route: <sip:192.168.1.237:5061;lr>
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 417
v=0
o=doubango 1983 678901 IN IP4 192.168.1.199
s=-
c=IN IP4 192.168.1.199
t=0 0
m=audio 65534 RTP/AVPF 9 101
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=acfg:1 t=1
a=sendrecv
a=rtcp-mux
a=ssrc:2174826863 cname:ldjWoB60jbyQlR6e
a=ssrc:2174826863 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2174826863 label:Doubango.Audio
==============================================
session.12: content-type=application/sdp plugin=com.brekeke.net.content.application.Sdp
session.12: pkt=6 dp=2 st=0 sip:700@192.168.1.237:5061(192.168.1.199:53611) --> sip:668@59.90.246.89(59.90.246.89:14334)
send="SIP/2.0 200 OK"
session.12: phase=4: Accepted
session.12: processtime=0
session.12: send: to=UAC:59.90.246.89:14334(UDP) at 04/10/13 18:27:08.531
==============================================
SIP/2.0 200 OK
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@59.90.246.89>;tag=1496733710
Contact: <sip:700@192.168.1.237:5061>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 INVITE
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=14334;received=59.90.246.89;branch=z9hG4bK1751344313
Record-Route: <sip:192.168.1.237:5061;lr>
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 417
v=0
o=doubango 1983 678901 IN IP4 192.168.1.237
s=-
c=IN IP4 192.168.1.237
t=0 0
m=audio 10002 RTP/AVPF 9 101
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=acfg:1 t=1
a=sendrecv
a=rtcp-mux
a=ssrc:2174826863 cname:ldjWoB60jbyQlR6e
a=ssrc:2174826863 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2174826863 label:Doubango.Audio
==============================================
session.12: receive: from=UAC:192.168.1.226:36306(UDP) at 04/10/13 18:27:08.625
==============================================
ACK sip:700@192.168.1.237:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.226:36306;branch=z9hG4bK486036623;rport=36306
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@59.90.246.89>;tag=1496733710
Contact: <sip:668@192.168.1.226:36306;transport=udp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 ACK
Max-Forwards: 70
Route: <sip:192.168.1.237:5061;lr>
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.532.758 (doubango r758 - Micromax A110)
Content-Length: 0
==============================================
session.12: pkt=7 dp=1 st=0 sip:668@59.90.246.89(192.168.1.226:36306) --> sip:700@192.168.1.237:5061(192.168.1.199:53611)
send="ACK sip:700@192.168.1.199:53611;transport=udp SIP/2.0"
session.12: phase=5: Talking
session.12: processtime=0
session.12: send: to=UAS:192.168.1.199:53611(UDP) at 04/10/13 18:27:08.625
==============================================
ACK sip:700@192.168.1.199:53611;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bK21233e0f205ac-30-2d73c1a
Via: SIP/2.0/UDP 192.168.1.226:36306;branch=z9hG4bK486036623;rport=36306
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>;tag=1496733710
Contact: <sip:668@192.168.1.237:5061>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223844 ACK
Max-Forwards: 69
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.532.758 (doubango r758 - Micromax A110)
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Length: 0
==============================================
session.12: receive: from=UAC:192.168.1.226:36306(UDP) at 04/10/13 18:27:19.296
==============================================
BYE sip:700@192.168.1.237:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.226:36306;branch=z9hG4bK309144096;rport=36306
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@59.90.246.89>;tag=1496733710
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223846 BYE
Max-Forwards: 70
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Route: <sip:192.168.1.237:5061;lr>
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.532.758 (doubango r758 - Micromax A110)
P-Preferred-Identity: <sip:668@59.90.246.89>
Content-Length: 0
==============================================
session.12: pkt=8 dp=1 st=0 sip:668@59.90.246.89(192.168.1.226:36306) --> sip:700@192.168.1.237:5061(192.168.1.199:53611)
send="BYE sip:700@192.168.1.199:53611;transport=udp SIP/2.0"
session.12: phase=6: Closing
session.12: processtime=0
session.12: send: to=UAS:192.168.1.199:53611(UDP) at 04/10/13 18:27:19.296
==============================================
BYE sip:700@192.168.1.199:53611;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bKee4a0b93205ac-30-2d73c1a
Via: SIP/2.0/UDP 192.168.1.226:36306;branch=z9hG4bK309144096;rport=36306
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>;tag=1496733710
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223846 BYE
Max-Forwards: 69
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.532.758 (doubango r758 - Micromax A110)
P-Preferred-Identity: <sip:668@59.90.246.89>
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Length: 0
==============================================
session.12: receive: from=UAS:192.168.1.199:53611(UDP) at 04/10/13 18:27:19.312
==============================================
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.237:5061;branch=z9hG4bKee4a0b93205ac-30-2d73c1a
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@192.168.1.237:5061>;tag=1496733710
Contact: <sip:700@192.168.1.199:53611;transport=udp>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223846 BYE
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=36306;branch=z9hG4bK309144096
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Length: 0
==============================================
session.12: pkt=9 dp=2 st=0 sip:700@192.168.1.237:5061(192.168.1.199:53611) --> sip:668@59.90.246.89(192.168.1.226:36306)
send="SIP/2.0 200 OK"
session.12: processtime=0
session.12: send: to=UAC:192.168.1.226:36306(UDP) at 04/10/13 18:27:19.312
==============================================
SIP/2.0 200 OK
From: <sip:668@59.90.246.89>;tag=1561177022
To: <sip:700@59.90.246.89>;tag=1496733710
Contact: <sip:700@192.168.1.237:5061>
Call-ID: 99993775-b7ac-0e8e-7551-ce8a87452f61
CSeq: 223846 BYE
Via: SIP/2.0/UDP 192.168.1.226:36306;rport=36306;branch=z9hG4bK309144096
Record-Route: <sip:192.168.1.237:5061;lr>
Content-Length: 0
==============================================
session.12: sipex.11: close: result=Success length=00:00:10 total-pkt=9 at 04/10/13 18:27:19.312
What kind of environment is it?
Are SIP Server and SIP clients located in the same LAN behind NAT?
It seems you set 59.90.246.89 as the SIP Server's interface IP address.
But SIP clients are also using the same global IP.
Why are you using 5061 for SIP Server?
Let you use 5060 because 5061 is reserved for SIP over TLS.
Are SIP Server and SIP clients located in the same LAN behind NAT?
It seems you set 59.90.246.89 as the SIP Server's interface IP address.
But SIP clients are also using the same global IP.
Why are you using 5061 for SIP Server?
Let you use 5060 because 5061 is reserved for SIP over TLS.
Since it looks SIP clients are in the same LAN, the SIP Server puts the local IP address instead of the global IP address.
If you set the following in the [Configuration]->[Advanced]page, the SIP Server will put the global IP address in SDP.
-----------------------------------
net.rtp.ifsrc=59.90.246.89
net.rtp.ifdst=59.90.246.89
-----------------------------------
You need to restart the SIP Server after you put them.
If you set the following in the [Configuration]->[Advanced]page, the SIP Server will put the global IP address in SDP.
-----------------------------------
net.rtp.ifsrc=59.90.246.89
net.rtp.ifdst=59.90.246.89
-----------------------------------
You need to restart the SIP Server after you put them.
>> Are SIP Server and SIP clients located in the same LAN behind NAT?
yes, sip clients and sip server are located in same LAN behind NAT.
we are registering the client using public IP. because the clients are running in andorid mobile.so the user may use local WIFI or GPRS network.so this IP address we should not exposed to the user to change.
>> Why are you using 5061 for SIP Server?
since we are using GPRS network in mobile,some operators blocked 5060 port.to avoid that we are using 5061.
yes, sip clients and sip server are located in same LAN behind NAT.
we are registering the client using public IP. because the clients are running in andorid mobile.so the user may use local WIFI or GPRS network.so this IP address we should not exposed to the user to change.
>> Why are you using 5061 for SIP Server?
since we are using GPRS network in mobile,some operators blocked 5060 port.to avoid that we are using 5061.
i have checked the same as you suggested instead of advance page,if i put dial plan like this
$request = ^INVITE $ifsrc = 59.90.246.89
$ifdst = 59.90.246.89
public IP replaced,and audio works both side now.
and i am facing one more problem in audio. if i make a call A - > B audio works.
if i call B -> A both client RTP reached to sip server. sip server not delivering it to clients.
i have analyzed the difference is if 200OK with SDP contain RTP/AVPF audio not works.
if 200OK with SDP contain RTP/AVP audio works both side. i am not sure this sip client bug or its normal.and also not sure the problem with only RTP/AVPF.can you help me how to resolve this?
$request = ^INVITE $ifsrc = 59.90.246.89
$ifdst = 59.90.246.89
public IP replaced,and audio works both side now.
and i am facing one more problem in audio. if i make a call A - > B audio works.
if i call B -> A both client RTP reached to sip server. sip server not delivering it to clients.
i have analyzed the difference is if 200OK with SDP contain RTP/AVPF audio not works.
if 200OK with SDP contain RTP/AVP audio works both side. i am not sure this sip client bug or its normal.and also not sure the problem with only RTP/AVPF.can you help me how to resolve this?
If you put these net.rtp.ifsrc and net.rtp.ifdst in DialPlan, write the rule like the following.
[Matching Pattern]
$request = ^INVITE
[Deploy Pattern]
&net.rtp.ifsrc = 59.90.246.89
&net.rtp.ifdst = 59.90.246.89
$ifsrc and $ifdst are for defining an interface IP addresses in SIP header.
&net.rtp.ifsrc and &net.rtp.ifdst are or defining an interface IP addresses in SDP.
[Matching Pattern]
$request = ^INVITE
[Deploy Pattern]
&net.rtp.ifsrc = 59.90.246.89
&net.rtp.ifdst = 59.90.246.89
$ifsrc and $ifdst are for defining an interface IP addresses in SIP header.
&net.rtp.ifsrc and &net.rtp.ifdst are or defining an interface IP addresses in SDP.
if i use dial plan &net.rtp.ifsrc = 59.90.246.89 ,&net.rtp.ifdst = 59.90.246.89 like this, still it does't replace to global ip address.
For RTP/AVPF problem already i have pasted the SIP trace in above conversation.in that you can find invite will have RTP/AVP and response will be RTP/AVPF.
i have not specifically enable or disable this AVPF in settings and the client does't have such option like this.
and i have tested with asterisk with same client.in this invite and response SDP having only RTP/AVP.
any configuration can be made for this in brekeke sip server or this is sip client fault?.
For RTP/AVPF problem already i have pasted the SIP trace in above conversation.in that you can find invite will have RTP/AVP and response will be RTP/AVPF.
i have not specifically enable or disable this AVPF in settings and the client does't have such option like this.
and i have tested with asterisk with same client.in this invite and response SDP having only RTP/AVP.
any configuration can be made for this in brekeke sip server or this is sip client fault?.