MediaProxy do not add Relay Candidate

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MediaProxy do not add Relay Candidate

vivid333
Finally I debug opensips, set breakpoint on use_media_proxy, found
MediaProxy Module don't find ice parameter(such as has_ice,
first_ice_candidate is not set). when I set ice param at the end of SDP,
then Relay works!

ie:

v=0
o=- 3531651689 3531651689 IN IP4 114.246.xxx.xxx
s=linphone
c=IN IP4 114.246.xxx.xxx
t=0 0
m=audio 46909 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
a=ice-ufrag:18e62b54
a=ice-pwd:5cf02fda
a=candidate:Sa00020f 1 UDP 1862270975 114.246.xxx.xxx 46909 typ srflx raddr 10.0.2.15 rport 26836
a=candidate:Ha00020f 1 UDP 1694498815 10.0.2.15 26836 typ host







On 2011年12月01日 19:00, [hidden email] wrote:

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> Today's Topics:
>
>     1. Re: OpenSIPS as the Firewall (Nick Khamis)
>     2. MediaProxy + OpenSIPS Integration (Nick Khamis)
>     3. MediaProxy do not add Relay Candidate (vivid333)
>     4. About SIP Traffic (Nick)
>     5. cdr accounting on opensips restart (Jayesh Nambiar)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 30 Nov 2011 19:26:59 -0500
> From: Nick Khamis<[hidden email]>
> Subject: Re: [OpenSIPS-Users] OpenSIPS as the Firewall
> To: OpenSIPS users mailling list<[hidden email]>
> Message-ID:
> <CAGWRaZY+=[hidden email]>
> Content-Type: text/plain; charset=ISO-8859-1
>
> MediaProxy it is! Thank you sir.
>
> Cheers,
>
> Nick
>
> On Wed, Nov 30, 2011 at 7:15 PM,<[hidden email]>  wrote:
>> I use Mediaproxy and don't have any issues with it. I have no experience
>> with RTPProxy. OpenSIPS b2bua is not what you are looking for when it comes
>> to RTP. The B2B_LOGIC module can do network hiding (b2b_init_request("top
>> hiding")) but so can the dialog module now (topology_hiding()).
>>
>> With Mediaproxy you can just add more Mediaproxies if the first one is
>> getting used too much. So from a loadbalancing and HA perspective Mediaproxy
>> is very simple to deploy. It also supports ICE so that could be good
>> depending on your clients.
>>
>> This tutorial for Mediaproxy is pretty old but should help you get started
>> http://voiprookie.blogspot.com/2009/04/blog-post.html
>>
>>
>>
>>
>>
>> On , Nick Khamis<[hidden email]>  wrote:
>>> Hey Duane,
>>>
>>>
>>>
>>>
>>>
>>> Thank you so much for your response. That is exactly my problem.
>>>
>>>
>>> Currently I only have
>>>
>>>
>>> OpenSIPS flowing SIP packets.
>>>
>>>
>>>
>>>
>>>
>>> As for the actual RTP, I was thinking of using either the STUN or
>>>
>>>
>>> NATHELPER module.
>>>
>>>
>>> Only I am not worried about NAT, I only need the flowing of RTP along
>>>
>>>
>>> with SIP. What
>>>
>>>
>>> is the most lightweight, and elgant solution to flow RTP? RTP Proxy
>>>
>>>
>>> from b2bua.org?
>>>
>>>
>>> I read somwhere that OpenSIPS also has a network hiding, and b2bua
>>>
>>>
>>> layer, is this
>>>
>>>
>>> the silver bullit I am looking for?
>>>
>>>
>>>
>>>
>>>
>>> Thanks in Advnace,
>>>
>>>
>>>
>>>
>>>
>>> Nick
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Nov 30, 2011 at 5:08 PM, ?[hidden email]>  wrote:
>>>
>>>
>>>> Are your diagrams the path that SIP takes or RTP? Asterisk can be
>>>> internal
>>>
>>>> and private from the outside world when it comes to SIP. It can also be
>>>
>>>> internal only when it comes to RTP but you would need to use a relay
>>>> server
>>>
>>>> like Mediaproxy or RTPProxy. Mediaproxy can sit on the internet with a
>>>
>>>> public IP and Asterisk can be behind a firewall.
>>>
>>>
>>>
>>>
>>>
>>>> On , Nick Khamis [hidden email]>  wrote:
>>>
>>>>> Hello Everyone,
>>>
>>>
>>>
>>>
>>>>> We are trying to close the doors entirely to our asterisk servers,
>>>>> making
>>>
>>>
>>>>> only opensips visible to the outside world:
>>>
>>>
>>>
>>>
>>>>> Incoming ?->  OpenSIPS ->  Asterisk ->  OpenSIPS ->  Trunk
>>>
>>>
>>>>> ? ? ? ? ? ? ? | ? ? ? ? ? In ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? Out
>>>
>>>>> ? |
>>>
>>>
>>>>> ? ? ? ? ? ? ? | _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ |
>>>
>>>
>>>
>>>
>>>
>>>
>>>>> What I think we currently have is:
>>>
>>>
>>>
>>>
>>>>> Incoming ?->  OpenSIPS ->  Asterisk ->  OpenSIPS ->  Trunk
>>>
>>>
>>>>> ? ? ? ? ? ? ? ? ? ? ? ?In ? ? ? ? ? ? ? ? ?| ? ? ? ? ? ? ? ?Out
>>>
>>>>> ? ?|
>>>
>>>
>>>>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?| _ _ _ _ _ _ _ _ _ __ _ _ |
>>>
>>>
>>>
>>>
>>>>> Without any port forwarding to the OpenSIPS box, everything
>>>
>>>
>>>>> works fine. With port forwarding, I get no audio both ways.
>>>
>>>
>>>
>>>
>>>>> If I am not mistaken, my questions are:
>>>
>>>
>>>>> * Can this be achieved
>>>
>>>
>>>>> * Do we have an externip, and port range settings for OpenSIPS.
>>>
>>>
>>>
>>>
>>>>> Thanks in Advnace,
>>>
>>>
>>>
>>>
>>>>> Nick
>>>
>>>
>>>
>>>
>>>>> _______________________________________________
>>>
>>>
>>>>> Users mailing list
>>>
>>>
>>>>> [hidden email]
>>>
>>>
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>>> _______________________________________________
>>>
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>>>
>>>
>>>
>>>
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>
>
> ------------------------------
>
> Message: 2
> Date: Wed, 30 Nov 2011 22:17:16 -0500
> From: Nick Khamis<[hidden email]>
> Subject: [OpenSIPS-Users] MediaProxy + OpenSIPS Integration
> To: OpenSIPS users mailling list<[hidden email]>
> Message-ID:
> <CAGWRaZaNxGEQV+Kq52zoYU=XjQ6qe1=8-BpuY_UL08SLda6=[hidden email]>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hello Everyone,
>
> We have been successful so far with our virutal machine prototpe:
>
> 2 VM for HA Mysql
> 2 VM for Asterisk
> 2 VM for OpenSIPS (in/out)
>
> Now we would like to iinclude MediaProxy with the OpenSIPS-in VM
> however, the webiste mentioned not to install MediaProxy on a VM?
> This is just a protoype, and will be installed on the host once the entire
> architecture is complete, configs etc..
> We want to compile it from source. Do you feel this will be a problem,
> and not function as expected? Good documentation from install to
> configuration of MediaProxy + OpenSIPS would be greatly appreciated.
>
> Thanks in Advnace,
>
> Nick.
>
>
>
> ------------------------------
>
> Message: 3
> Date: Thu, 01 Dec 2011 15:09:03 +0800
> From: vivid333<[hidden email]>
> Subject: [OpenSIPS-Users] MediaProxy do not add Relay Candidate
> To: [hidden email]
> Message-ID:<[hidden email]>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hello:
>
>               A->opensips (+mediaproxy)->  B
>
>               Can any one help me why MediaProxy don't add Relay
> candidate? I am sure that use_media_proxy works before  opensips send
> invite to B.
>
>
> ////////////////////////////A(pjsip) Send Invite,  B side receive Invite
> include relay candidate
> *********************************************************************************************
> INVITE sip:[hidden email]:5060 SIP/2.0
> Via: SIP/2.0/UDP
> 114.249.xxx.xxx:32768;rport;branch=z9hG4bKPjRJalhnJGIbBwiSvzyqUo2rgOM0ld3Wqo
> Max-Forwards: 70
> From: sip:+[hidden email];tag=qv4WrTe2HkdTK8U2W8Y40SgAeHIehYGR
> To: sip:[hidden email]
> Contact:<sip:+[hidden email]:32768;ob>
> Call-ID: xxAb8TPG-d-ePhIOx1Xw6jyDjywCJDeU
> CSeq: 23079 INVITE
> Route:<sip:225.4.xxx.xxx:5060;lr>
> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
> REFER, MESSAGE, OPTIONS
> Supported: replaces, 100rel, timer, norefersub
> Session-Expires: 1800
> Min-SE: 90
> User-Agent: PJSUA v1.10.0 Linux-2.6.32.35/i686/glibc-2.11
> Content-Type: application/sdp
> Content-Length:   629
>
> v=0
> o=- 3531703180 3531703180 IN IP4 114.246.xxx.xxx
> s=pjmedia
> c=IN IP4 114.246.xxx.xxx
> t=0 0
> a=X-nat:7
> m=audio 56846 RTP/AVP 98 97 99 104 3 0 8 9 96
> a=rtpmap:98 speex/16000
> a=rtpmap:97 speex/8000
> a=rtpmap:99 speex/32000
> a=rtpmap:104 iLBC/8000
> a=fmtp:104 mode=30
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:9 G722/8000
> a=sendrecv
> a=rtpmap:96 telephone-event/8000
> a=fmtp:96 0-15
> a=ice-ufrag:025fe5d4
> a=ice-pwd:72314495
> a=candidate:Sa00020f 1 UDP 1862270975 114.246.xxx.xxx 56846 typ srflx
> raddr 10.0.2.15 rport 49367
> a=candidate:Ha00020f 1 UDP 1694498815 10.0.2.15 49367 typ host
>
>
> ////////////////////////////A(linphone). change Pjsip to linphone,  B
> side receive Invite have no relay candidate
> *********************************************************************************************
> INVITE sip:[hidden email] SIP/2.0
> Via: SIP/2.0/UDP 10.0.2.15:41338;rport;branch=z9hG4bK2713821050
> From:<sip:[hidden email]>;tag=2642865755
> To: "TEST"<sip:[hidden email]>
> Call-ID: 1915536026
> CSeq: 20 INVITE
> Contact:<sip:vivid333@10.0.2.15:41338>
> Content-Type: application/sdp
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO
> Max-Forwards: 70
> User-Agent: Linphone
> Supported: replaces, 100rel, timer, norefersub
> Content-Length:   413
>
> v=0
> o=- 3531651689 3531651689 IN IP4 114.246.xxx.xxx
> s=linphone
> c=IN IP4 114.246.xxx.xxx
> t=0 0
> a=ice-ufrag:18e62b54
> a=ice-pwd:5cf02fda
> a=candidate:Sa00020f 1 UDP 1862270975 114.246.xxx.xxx 46909 typ srflx
> raddr 10.0.2.15 rport 26836
> a=candidate:Ha00020f 1 UDP 1694498815 10.0.2.15 26836 typ host
> m=audio 46909 RTP/AVP 0 96
> a=rtpmap:0 PCMU/8000
> a=rtpmap:96 telephone-event/8000
> a=fmtp:96 0-15
> a=sendrecv
>
>
>
>
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Thu, 01 Dec 2011 17:38:28 +0800
> From: Nick<[hidden email]>
> Subject: [OpenSIPS-Users] About SIP Traffic
> To: OpenSIPS users mailling list<[hidden email]>
> Message-ID:<[hidden email]>
> Content-Type: text/plain; charset=UTF-8; format=flowed
>
> Hi
>
> I want to monitor SIP Traffic with MRTG.
> But, I don't want to monitor network card. In Server, I have other service.
>
> Do everyone have other function with it??
>
> Can snmpstats support this??  Which function can do it??
>
> Thanks for suggest.
> Nick
>
>
>
>
>
> ------------------------------
>
> Message: 5
> Date: Thu, 1 Dec 2011 15:58:22 +0530
> From: Jayesh Nambiar<[hidden email]>
> Subject: [OpenSIPS-Users] cdr accounting on opensips restart
> To: OpenSIPS users mailling list<[hidden email]>
> Message-ID:
> <[hidden email]>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hello All,
> I am planning to use CDR accounting in my script starting from version 1.7
> and it looks fine and working as expected. Although I had one doubt, how do
> I make sure the CDR accounting still happens if the opensips is restarted
> and BYE comes after the restart. I have tried db_mode 3 for dialog module
> so that it dumps all the dialogs while shutdown and on start it fetches the
> dialog from the DB. This method makes sure the dialog is matched when BYE
> comes after the restart but the CDR record is not entered.
> Is there any flag or dialog variable that I should set to insert that value
> in the table for all dialogs when opensips shuts off so that opensips knows
> that the CDR flag was set for this dialog when started again and it has to
> insert the record?
> Do let me know for any pointers or ideas to get this done.
> Thanks in advance.
>
> --- Jayesh
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