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, [email protected] 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<[email protected]>
Subject: Re: [OpenSIPS-Users] OpenSIPS as the Firewall
To: OpenSIPS users mailling list<[email protected]>
Message-ID:
<CAGWRaZY+=tkrog4-xxnp6zcbcw0myjhvevyqjdp50fmcs0+...@mail.gmail.com>
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,<[email protected]> 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<[email protected]> 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, [email protected]> 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 [email protected]> 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
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------------------------------
Message: 2
Date: Wed, 30 Nov 2011 22:17:16 -0500
From: Nick Khamis<[email protected]>
Subject: [OpenSIPS-Users] MediaProxy + OpenSIPS Integration
To: OpenSIPS users mailling list<[email protected]>
Message-ID:
<CAGWRaZaNxGEQV+Kq52zoYU=XjQ6qe1=8-BpuY_UL08SLda6=g...@mail.gmail.com>
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<[email protected]>
Subject: [OpenSIPS-Users] MediaProxy do not add Relay Candidate
To: [email protected]
Message-ID:<[email protected]>
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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP
114.249.xxx.xxx:32768;rport;branch=z9hG4bKPjRJalhnJGIbBwiSvzyqUo2rgOM0ld3Wqo
Max-Forwards: 70
From: sip:[email protected];tag=qv4WrTe2HkdTK8U2W8Y40SgAeHIehYGR
To: sip:[email protected]
Contact:<sip:[email protected]: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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.2.15:41338;rport;branch=z9hG4bK2713821050
From:<sip:[email protected]>;tag=2642865755
To: "TEST"<sip:[email protected]>
Call-ID: 1915536026
CSeq: 20 INVITE
Contact:<sip:[email protected]: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<[email protected]>
Subject: [OpenSIPS-Users] About SIP Traffic
To: OpenSIPS users mailling list<[email protected]>
Message-ID:<[email protected]>
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<[email protected]>
Subject: [OpenSIPS-Users] cdr accounting on opensips restart
To: OpenSIPS users mailling list<[email protected]>
Message-ID:
<calvf6vcj7bfuzoni_vlkzfpeyq-5exwboaz1jbgewpwjzop...@mail.gmail.com>
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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