[SR-Users] kamailio with mediaproxy-ng, 488 Not Acceptable Here

[Muhammad Shahzad]

There are several problems that need to be addressed in your kamailio.cfg
but let me try to focus only on mediaprxoy-ng related ones.

First instead of engaging mediaproxy in failure route, engage it main route
or branch route. Why wait for failure when we know call will fail anyway if
you try to call webrtc to sip or vice versa.

Secondly you need to keep track of connection type of both caller and
callee and set appropriate mediaproxy-ng flags according to call direction,
e.g. call from webrtc to sip, or sip to webrtc or webrtc to webrtc or sip
to sip, each type of call needs different set of flags for both
rtpproxy_offer and rtpproxy_answer.

How you do this, is pretty simple, to detect if caller is webrtc endpoint
you can use,

if ($avp(mline) =~ "SAVPF") {
# caller is a webrtc endpoint
};

To check if callee is a webrtc endpoint, you can use,

if ($(ru{uri.param,transport}) =~ "ws") {
# callee is a webrtc endpoint
};

For testing purpose, i recommend you only use mediaproxy-ng for bridging
webrtc to sip or vice versa calls, i.e. if both endpoints are using same
transport (e.g. sip to sip or webrtc to webrtc calls) then don't use
mediaproxy-ng at all and allow endpoints to establish media directly (that
would work out the box at least for webrtc to webrtc calls).

Finally use correct flags for each type of call (i recommend doing it in
branch route), for example,

For WebRTC to SIP call use flags (case-sensitive),

$avp(rtpproxy_offer_flags)  = "froc-sp";
$avp(rtpproxy_answer_flags) = "froc+SP";
rtpproxy_offer($avp(rtpproxy_offer_flags));

For SIP to WebRTC call use flags (case-sensitive),

$avp(rtpproxy_offer_flags)  = "froc+SP";
$avp(rtpproxy_answer_flags) = "froc-sp";
rtpproxy_offer($avp(rtpproxy_offer_flags));


Then in reply route,

rtpproxy_answer($avp(rtpproxy_answer_flags));


Remember, currently mediaproxy-ng does NOT support SRTP/DTLS, which is
required by firefox, so as result your webrtc endpoint MUST be running on
Chrome.

Hope this helps.

Thank you.




On Tue, Feb 4, 2014 at 3:28 PM, Mihai Marin <marinmihai at gmail.com> wrote:

> Hello,
> Thank you for your support.
>
> Yes, I have the same error without video enabled. I have attached the logs
> from jssip (with and without video support) and logs from kamailio when
> trying a call with video support enabled. The kamailio.cfg used is the same
> from my previous mail.
>
> I also tried with sipml5 and I have the same behavior.
>
> I'm stuck on this error and I think I'm looking in the wrong direction.
>
> Thank you.
>
> Best regards,
> Mihai M
>
>
> On Tue, Feb 4, 2014 at 2:49 PM, Andrew Pogrebennyk <
> apogrebennyk at sipwise.com> wrote:
>
>> Hi,
>> could you please post also your Chrome js developer log?
>> Does the problem exist if you start the jssip clients without video
>> support?
>>
>> Andrew
>>
>> On 02/03/2014 12:00 PM, Mihai Marin wrote:
>> > Hello,
>> >
>> > Another weekend struggling to make a call from jssip to another jssip
>> > behind firewall and I still receive 488 - Not Acceptable Here. I tried
>> > all the ideas that I had/received without any success - including catch
>> > 488 and re-invite.
>> > [...]
>> > What do I miss from my configuration?
>> >
>> > Thank you.
>> >
>> > Best regards,
>> > Mihai M
>>
>>
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>
>
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