[SR-Users] How does rtpproxy handle handover?

[SamyGo]

Dear Khoa,

As Daniel stated you need to see if your SIP phones are able to sense the
change in its network parameters and trigger a Re-INVITE to Kamailio with
new SDP to handle the audio.

That's very important to do because once RTPproxy allocates the ports it
can't just start sending RTPs to the new A2 network IP on its own OR start
receiving RTPs from new IP to its already allocated port (that'll mean
anyone can send RTPs to an allocated port and insert media into someone
else's call).

Please do share the network topology where the first network transition
worked, possibly it's Public IP remained the same and maybe your internal
network handled that somehow(NAT/PAT) !??

Now once the Re-INVITES are exchanged only then RTPproxy will be explored
to see if it handles the transparent Handover/updates or not.

BR,
Sammy






On Wed, Aug 14, 2013 at 7:03 AM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>  Kamailio does not send any command to RTPProxy unless it handles some SIP
> messages, the U and L commands are typically for INVITEs and 200ok.
>
> Have you looked at sip traffic? You can run ngrep on kamailio server:
>
> ngrep -d any -qt -W byline port 5060
>
> Cheers,
> Daniel
>
>
> On 8/14/13 1:40 PM, Khoa Pham wrote:
>
> I think it is related to so called IP address filling and trusted IP
>
>
> On Wed, Aug 14, 2013 at 4:09 PM, Khoa Pham <onmyway133 at gmail.com> wrote:
>
>> Hi Daniel,
>>
>>  My clients don't do anything when IP change occurs.From what I inspect,
>> it is because of rtpproxy does not accept the 2nd IP change.
>> The the rtpproxy protocol document
>> http://www.rtpproxy.org/wiki/RTPproxy/Protocol, the Update and Lookup
>> command have [arg] parameters.
>> U[args] callid addr port from_tag [to_tag [notify_socket [notify_args]]]
>>  L[args] callid addr port from_tag to_tag
>>
>>  I see Kamailio often send Uc and Lc to rtpproxy. I still can't find out
>> what these arg mean, but maybe it's the point
>>
>>
>>  On Wed, Aug 14, 2013 at 3:31 PM, Daniel-Constantin Mierla <
>> miconda at gmail.com> wrote:
>>
>>>   Hello,
>>>
>>>
>>> On 8/13/13 5:56 AM, Khoa Pham wrote:
>>>
>>>  I have SIP proxy (Kamailio) works in conjunction with rtpproxy<http://www.rtpproxy.org/> to
>>> support client communication. When SIP proxy sends command to rtpproxy to
>>> create new session, rtpproxy will create 2 ports (let's called them port1
>>> and port2). rtpproxy has 1 listen interface
>>>
>>> Supposed A and B are 2 clients that use rtpproxy to relay RTP stream,
>>> and works fine.
>>>
>>> A <---> port1 [*rtpproxy*] port2 <---> B
>>>
>>>  Now that A loses his current network, and enter network2 (imagine a
>>> network handover) to become A2. In this case, I see rtpproxy still works
>>> fine by relaying stream between A2 and B
>>>
>>> A2 <---> port1 [*rtpproxy*] port2 <---> B
>>>
>>>  But when A2 lose his network2 and enters network3 to become A3,
>>> rtpproxy stills relay stream between A2 and B. It seems that A can change
>>> his network only once.
>>>
>>> A2 <---> port1 [*rtpproxy*] port2 <---> B
>>>
>>> A3
>>>
>>>  Why did the first handover succeed? How can I change rtpproxy behavior
>>> to support many handovers ?
>>>
>>>  what I expect that happened between A and A2 is that the client
>>> application sent a re-INVITE with its new IP address. But then it didn't
>>> happen when going to A3. Rtpproxy itself can do nothing here. You should
>>> look at sip traffic to see what happens.
>>>
>>> Cheers,
>>> Daniel
>>>
>>> --
>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>>
>>>
>>>  _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>>
>>  --
>> Khoa Pham
>> HCMC University of Science
>> www.fantageek.com
>>
>
>
>
>  --
> Khoa Pham
> HCMC University of Science
> www.fantageek.com
>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20130814/6cb35b13/attachment.html>