5 Feb
2014
5 Feb
'14
10:41 a.m.
Hey,
If you're trying to connect two WebRTC endpoints with each, you don't
need any of mediaproxy-ng's magic to get it working. All the previous
replies were assuming that you were trying to connect a WebRTC endpoint
with a non-WebRTC one, which is usually what people are trying to do.
In your case, the flags "froc" in either direction should be sufficient
to get the job done. If it still doesn't work, can you please post the
rejected SDP body as it appears both on the sender's side and on the
receiver's side (i.e. both before and after it went through MP-NG).
cheers
On 02/05/14 05:17, Mihai Marin wrote:
Hello,
Thank you for your detailed explication but I miss some information or
I'm unable to understand it properly. What I'm trying to do is to use
mediaproxy-ng as a turn server between 2 WebRTC endpoints (when at least
one is behind restrictive firewall). Trying to replicate what you
explained on my needs I tried:
$avp(rtpproxy_offer_flags) = "froc+SP";
$avp(rtpproxy_answer_flags) = "froc-SP";
But, unfortunately, I have the same error. Sorry if the solution is
obvious but I can't find it.
Thank you.
Best regards,
Mihai M
On Tue, Feb 4, 2014 at 10:45 PM, Muhammad Shahzad <shaheryarkh(a)gmail.com
<mailto:shaheryarkh@gmail.com>> wrote:
There are several problems that need to be addressed in your
kamailio.cfg but let me try to focus only on mediaprxoy-ng related ones.
First instead of engaging mediaproxy in failure route, engage it
main route or branch route. Why wait for failure when we know call
will fail anyway if you try to call webrtc to sip or vice versa.
Secondly you need to keep track of connection type of both caller
and callee and set appropriate mediaproxy-ng flags according to call
direction, e.g. call from webrtc to sip, or sip to webrtc or webrtc
to webrtc or sip to sip, each type of call needs different set of
flags for both rtpproxy_offer and rtpproxy_answer.
How you do this, is pretty simple, to detect if caller is webrtc
endpoint you can use,
if ($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(a)gmail.com
<mailto:marinmihai@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(a)sipwise.com <mailto:apogrebennyk@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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