How to use a different port number for voip traffic

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Hi Fadi,

if your proxy wishes to remain on the path of requests within the 
established dialog, configure it to *record-route*.
It wants/loves traffic, it'll sure get some ;o)

Cheers,
Alain

On 31-Jan-12 16:11, Fadi Chehimi wrote:
> Hello again
>
> I am resending this email hoping that someone will be able to help.
>
> Cheers
> Fadi
>
>
> On Thu, Jan 19, 2012 at 11:43 AM, Fadi Chehimi<fchehimi at localphone.com>wrote:
>
>> Thanks Johan for this
>> I am now getting closer to the solution.
>> Now I the INVITES routed through the port I am after, but for some
>> reason ACK and BYE are routed through 5060 still. Not sure if I have
>> to manually edit port for these commands as well. I have attached here
>> some of the consol logs:
>>
>>
>> PJSIP initialization..
>> ..
>>   11:29:55.533   pjsua_core.c  SIP UDP socket reachable at
>> 10.15.20.147:12345
>> ...
>> ...
>> INVITE sip:9198542 at domainname.com SIP/2.0
>> Via: SIP/2.0/UDP
>> 10.15.20.147:12345;rport;branch=z9hG4bKPjm6ezkf5cgXbSZQA.3oc4JTABzqlfaTCj
>> Max-Forwards: 70
>> From: "FadiTest CheTest"
>> <sip:8932153 at domainname.com>;tag=0S4PJkPDy8oOArXzTNchLLQ8X19bCPaJ
>> To: sip:9198542 at domainname.com
>> Contact: "FadiTest CheTest"<sip:8932153 at 10.15.20.147:12345;ob>
>> Call-ID: bLCNgGoNpAq89tUEs7IMoS7LzmwMOGml
>> CSeq: 23363 INVITE
>> Route:<sip:gateway.domainname.com:12345;lr>
>> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
>> REFER, MESSAGE, OPTIONS
>> Supported: replaces, 100rel, timer, norefersub
>> Session-Expires: 1800
>> Min-SE: 300
>> User-Agent: domainname
>> Content-Type: application/sdp
>> Content-Length:   426
>> ...
>> ...
>>
>> --end msg--
>>   11:30:48.831 os_core_unix.c  Info: possibly re-registering existing thread
>>   11:30:48.834 callhandling.m  Call 0 state changed to CALLING
>> 2012-01-19 11:30:48.839 domainname2[4887:6f03] run loop
>>   11:30:49.330   pjsua_core.c  TX 1215 bytes Request msg
>> INVITE/cseq=23363 (tdta0xbb9200) to UDP 108.59.2.137:12345:
>> INVITE sip:9198542 at domainname.com SIP/2.0
>> Via: SIP/2.0/UDP
>> 10.15.20.147:12345;rport;branch=z9hG4bKPjm6ezkf5cgXbSZQA.3oc4JTABzqlfaTCj
>> Max-Forwards: 70
>> From: "FadiTest CheTest"
>> <sip:8932153 at domainname.com>;tag=0S4PJkPDy8oOArXzTNchLLQ8X19bCPaJ
>> To: sip:9198542 at domainname.com
>> Contact: "FadiTest CheTest"<sip:8932153 at 10.15.20.147:12345;ob>
>> Call-ID: bLCNgGoNpAq89tUEs7IMoS7LzmwMOGml
>> CSeq: 23363 INVITE
>> Route:<sip:gateway.domainname.com:12345;lr>
>> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
>> REFER, MESSAGE, OPTIONS
>> Supported: replaces, 100rel, timer, norefersub
>> Session-Expires: 1800
>> Min-SE: 300
>> User-Agent: domainname Sip, iPhone App v1.1.1
>> X-LP-GeoLocation: LONDON, gb
>> X-LP-Connection-Type: Wi-Fi
>> Content-Type: application/sdp
>> Content-Length:   426
>> ...
>> ...
>>
>> --end msg--
>>   11:30:52.464   pjsua_core.c  TX 444 bytes Request msg ACK/cseq=23363
>> (tdta0xbb5a00) to UDP 108.59.2.137:12345:
>> ACK sip:9198542 at domainname.com SIP/2.0
>> Via: SIP/2.0/UDP
>> 10.15.20.147:12345;rport;branch=z9hG4bKPjm6ezkf5cgXbSZQA.3oc4JTABzqlfaTCj
>> Max-Forwards: 70
>> From: "FadiTest CheTest"
>> <sip:8932153 at domainname.com>;tag=0S4PJkPDy8oOArXzTNchLLQ8X19bCPaJ
>> To: sip:9198542 at domainname.com;tag=4fdf1f17ba0fc0dcebac7ca3eba0e50d.dc1d
>> Call-ID: bLCNgGoNpAq89tUEs7IMoS7LzmwMOGml
>> CSeq: 23363 ACK
>> Route:<sip:gateway.domainname.com:12345;lr>
>> Content-Length:  0
>> ..
>> ..
>>
>> --end msg--
>>   11:30:52.482 callhandling.m  Call 0 state changed to CALLING
>>   11:30:52.666   pjsua_core.c  RX 400 bytes Response msg
>> 100/INVITE/cseq=23364 (rdata0xb7f014) from UDP 108.59.2.137:12345:
>> SIP/2.0 100 Giving a try
>> Via: SIP/2.0/UDP
>> 10.15.20.147:12345
>> ;received=188.39.51.2;rport=12345;branch=z9hG4bKPjAPy4b7DDpAGVeJ-Yg51x9xEXyeVttlXo
>> From: "FadiTest CheTest"
>> <sip:8932153 at domainname.com>;tag=0S4PJkPDy8oOArXzTNchLLQ8X19bCPaJ
>> To: sip:9198542 at domainname.com
>> Call-ID: bLCNgGoNpAq89tUEs7IMoS7LzmwMOGml
>> CSeq: 23364 INVITE
>> Server: OpenSIPS (1.7.1-notls (x86_64/linux))
>> Content-Length: 0
>> ...
>> ...
>>
>> --end msg--
>>   11:30:52.974   conference.c  Port 3 (ringback) transmitting to port 0
>> (iPhone IO device)
>>   11:30:52.974 callhandling.m  Call 0 state changed to EARLY (180 Ringing)
>>   11:30:57.869    udp0xb7de00  Remote RTP address switched to
>> 95.211.119.213:15278
>>   11:30:57.869    udp0xb7de00  Remote RTCP address switched to
>> 95.211.119.213:15279
>>   11:30:59.040   pjsua_core.c  RX 1137 bytes Response msg
>> 200/INVITE/cseq=23364 (rdata0xb7f014) from UDP 108.59.2.137:12345:
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP
>> 10.15.20.147:12345;rport;branch=z9hG4bKPjAPy4b7DDpAGVeJ-Yg51x9xEXyeVttlXo
>> From: "FadiTest CheTest"
>> <sip:8932153 at domainname.com>;tag=0S4PJkPDy8oOArXzTNchLLQ8X19bCPaJ
>> To: sip:9198542 at domainname.com;tag=9e4a4ZyK11XUa
>> Call-ID: bLCNgGoNpAq89tUEs7IMoS7LzmwMOGml
>> CSeq: 23364 INVITE
>> Contact:<sip:108.59.2.137:5060;did=de1.b746f8e6>    <======== THIS
>> HERE SHOULD BE 12345
>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-c64f475 2011-01-18 14-36-30
>> -0500
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>> Require: timer
>> Supported: timer, precondition, path, replaces
>> Allow-Events: talk, hold, refer
>> Session-Expires: 1800;refresher=uac
>> Min-SE: 300
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 272
>> Remote-Party-ID: "Outbound Call"
>> <sip:9198542 at 95.211.119.213>;party=calling;privacy=off;screen=no
>> ...
>> ...
>>
>> --end msg--
>>   11:30:59.055 callhandling.m  Call 0 state changed to CONNECTING
>> [Switching to thread 13315]
>>   11:30:59.598   strm0xbc31b4  VAD temporarily disabled
>>   11:30:59.601   strm0xbc31b4  Encoder stream started
>>   11:30:59.601   strm0xbc31b4  Decoder stream started
>>   11:30:59.609  pjsua_media.c  Media updates, stream #0: iLBC (sendrecv)
>>   11:30:59.611   conference.c  Port 3 (ringback) stop transmitting to
>> port 0 (iPhone IO device)
>>   11:30:59.613   conference.c  Port 4 (sip:9198542 at domainname.com)
>> transmitting to port 0 (iPhone IO device)
>>   11:30:59.613   conference.c  Port 0 (iPhone IO device) transmitting
>> to port 4 (sip:9198542 at domainname.com)
>>   11:30:59.615   pjsua_core.c  TX 386 bytes Request msg ACK/cseq=23364
>> (tdta0xede000) to UDP 108.59.2.137:5060:<======== THIS HERE SHOULD
>> BE 12345
>> ACK sip:108.59.2.137:5060;did=de1.b746f8e6 SIP/2.0
>> Via: SIP/2.0/UDP
>> 10.15.20.147:12345;rport;branch=z9hG4bKPjZNZiwUuuXAkSHBRRtjWNjIz2kRQ2tWH4
>> Max-Forwards: 70
>> From: "FadiTest CheTest"
>> <sip:8932153 at domainname.com>;tag=0S4PJkPDy8oOArXzTNchLLQ8X19bCPaJ
>> To: sip:9198542 at domainname.com;tag=9e4a4ZyK11XUa
>> Call-ID: bLCNgGoNpAq89tUEs7IMoS7LzmwMOGml
>> CSeq: 23364 ACK
>> Content-Length:  0
>>
>>
>> --end msg--
>>   11:30:59.618 callhandling.m  Call 0 state changed to CONFIRMED
>>   11:30:59.676   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
>>   11:31:00.260   strm0xbc31b4  VAD re-enabled
>>   11:31:05.303   conference.c  Port 0 (iPhone IO device) stop
>> transmitting to port 4 (sip:9198542 at domainname.com)
>>   11:31:05.303   conference.c  Port 2
>>
>> (/var/mobile/Applications/4E314999-EBC0-41E9-9587-136EE83AF2BB/domainname2.app/hang-up.wav)
>> transmitting to port 4 (sip:9198542 at domainname.com)
>>   11:31:06.304   conference.c  Port 2
>>
>> (/var/mobile/Applications/4E314999-EBC0-41E9-9587-136EE83AF2BB/domainname2.app/hang-up.wav)
>> stop transmitting to port 4 (sip:9198542 at domainname.com)
>>   11:31:06.304   conference.c  Port 0 (iPhone IO device) transmitting
>> to port 4 (sip:9198542 at domainname.com)
>>   11:31:06.305   pjsua_core.c  TX 433 bytes Request msg BYE/cseq=23365
>> (tdta0xee6000) to UDP 108.59.2.137:5060:<======== THIS HERE SHOULD
>> BE 12345
>> BYE sip:108.59.2.137:5060;did=de1.b746f8e6 SIP/2.0
>> Via: SIP/2.0/UDP
>> 10.15.20.147:12345;rport;branch=z9hG4bKPj.oGON31LnBSH9GNcVLHPJe4so5fG3LzD
>> Max-Forwards: 70
>> From: "FadiTest CheTest"
>> <sip:8932153 at domainname.com>;tag=0S4PJkPDy8oOArXzTNchLLQ8X19bCPaJ
>> To: sip:9198542 at domainname.com;tag=9e4a4ZyK11XUa
>> Call-ID: bLCNgGoNpAq89tUEs7IMoS7LzmwMOGml
>> CSeq: 23365 BYE
>> User-Agent: domainname Sip, iPhone App v1.1.1
>> Content-Length:  0
>> ...
>> ...
>>
>>
>>
>>> sipProxy = sip:ip:port in the pjsua config?
>>>
>>> BR
>>>
>>> Johan
>>>


-- 
                             ""
                           (o)(o)
                 _____o00o__(__)__o00o_____
3072D/146D10DE 2011-09-29    Alain Totouom  <totouom at gmx.de>
PGP Fingerprint 39A4F092 FFA7C746 CC305CB0 69091911 146D10DE



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