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bensly
Brekeke Addict
Joined: 02 Nov 2007
Posts: 27
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 Posted: Wed Apr 03, 2013 2:53 am Post subject: No voice Behind NAT |
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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 |
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lakeview
Brekeke Master Guru
Joined: 15 Nov 2007
Posts: 319
Location: Florida
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| Posted: Wed Apr 03, 2013 12:47 pm Post subject: |
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Which SIP client soft are you using on Android?
It seems the SIP client didn't put its global IP address in SDP.
So... let you tune the SIP Server to accept such a SIP client.
Go to the [Configuration]->[RTP] page, and set the [Send UA's remote address]="yes".
I hope it solves the issue. |
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bensly
Brekeke Addict
Joined: 02 Nov 2007
Posts: 27
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| Posted: Thu Apr 04, 2013 3:18 am Post subject: |
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Hi,
i am using imsdorid client in android mobile.as you said in SDP local ip address only sending by the client.but after enabled Send UA's remote address= yes also the problem is same.
Regards
Bensly |
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lakeview
Brekeke Master Guru
Joined: 15 Nov 2007
Posts: 319
Location: Florida
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| Posted: Thu Apr 04, 2013 10:57 am Post subject: |
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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-Brekeke-product-with-a-firewall |
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bensly
Brekeke Addict
Joined: 02 Nov 2007
Posts: 27
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| Posted: Mon Apr 08, 2013 6:56 am Post subject: |
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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 |
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redroof
Brekeke Talented
Joined: 16 Nov 2007
Posts: 97
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| Posted: Mon Apr 08, 2013 9:30 am Post subject: |
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> 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. |
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bensly
Brekeke Addict
Joined: 02 Nov 2007
Posts: 27
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| Posted: Tue Apr 09, 2013 12:27 am Post subject: |
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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 |
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redroof
Brekeke Talented
Joined: 16 Nov 2007
Posts: 97
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| Posted: Tue Apr 09, 2013 10:27 am Post subject: |
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Is 192.168.1.237 the SIP Server's local IP address?
Is 59.90.246.xx the SIP Server's global IP address?
Can you see the global IP address at the [Interface] field in the [Start/Shutdown] page? |
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bensly
Brekeke Addict
Joined: 02 Nov 2007
Posts: 27
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| Posted: Wed Apr 10, 2013 6:12 am Post subject: |
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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 |
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redroof
Brekeke Talented
Joined: 16 Nov 2007
Posts: 97
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| Posted: Wed Apr 10, 2013 10:53 pm Post subject: |
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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. |
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redroof
Brekeke Talented
Joined: 16 Nov 2007
Posts: 97
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| Posted: Wed Apr 10, 2013 11:58 pm Post subject: |
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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. |
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bensly
Brekeke Addict
Joined: 02 Nov 2007
Posts: 27
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| Posted: Wed Apr 10, 2013 11:58 pm Post subject: |
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>> 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. |
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bensly
Brekeke Addict
Joined: 02 Nov 2007
Posts: 27
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| Posted: Thu Apr 11, 2013 12:15 am Post subject: |
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I have tried as you suggested below in advance page
net.rtp.ifsrc=59.90.246.89
net.rtp.ifdst=59.90.246.89
but still it does't replace to public IP address. |
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bensly
Brekeke Addict
Joined: 02 Nov 2007
Posts: 27
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| Posted: Thu Apr 11, 2013 1:24 am Post subject: |
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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? |
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redroof
Brekeke Talented
Joined: 16 Nov 2007
Posts: 97
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| Posted: Thu Apr 11, 2013 10:22 am Post subject: |
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If you put these net.rtp.ifsrc and net.rtp.ifdst in DialPlan, write the rule like the following.
| Matching Patterns | $request = ^INVITE
| | Deploy Patterns | &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. |
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redroof
Brekeke Talented
Joined: 16 Nov 2007
Posts: 97
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| Posted: Thu Apr 11, 2013 10:48 am Post subject: |
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For RTP/AVPF's issue, what kind of info do you get in the [Session Details] page?
You can access to the [Session Details] page from the [Active Sessions] page during a call. |
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lakeview
Brekeke Master Guru
Joined: 15 Nov 2007
Posts: 319
Location: Florida
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| Posted: Thu Apr 11, 2013 2:36 pm Post subject: |
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bensly,
For your case, both SIP clients must enable RTP/AVPF instead of RTP/AVP. |
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bensly
Brekeke Addict
Joined: 02 Nov 2007
Posts: 27
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| Posted: Fri Apr 12, 2013 1:28 am Post subject: |
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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?. |
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