[SR-Users] How does rtpproxy handle handover?

[Khoa Pham]

Hi Daniel, Sammy, thanks for reply

But as Andres pointed out
http://lists.sip-router.org/pipermail/sr-users/2008-March/062795.html
After the pre-filling state, rtpproxy can re-fill media address. I think it
means that rtpproxy does not care about IP in U and L (Update, Lookup)
commands, it only cares about IP in the first RTP packet.

So in my case A2, it can re-fill 2 times.Look at the main.c in rtpproxy
source code, in the rxmit_packets() functions, it does the address filling

/*
* Update recorded address if it's necessary. Set "untrusted address"
* flag in the session state, so that possible future address updates
* from that client won't get address changed immediately to some
* bogus one.
*/



On Thu, Aug 15, 2013 at 1:22 PM, SamyGo <govoiper at gmail.com> wrote:

> 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
>>
>>
>


-- 
Khoa Pham
HCMC University of Science
www.fantageek.com
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