Hi all,
I am answering a SIP call made to my softphone. The softphone rings and also sends a 200 OK correctly. But the other end that made the call is seeing a failure in the SDP of the 200 OK packet.
This error is seen in SDP header => Invalid SDP line (no '=' delimiter) error
w.r.t w_frame4_public_ip.pcap It looks like the failure as seen in wireshark is because line "audio 5000 RTP/AVP 8 101" should really be "m= audio 5000 RTP/AVP 8 101"
so it doesn't see the '=' delimiter.
But if you look at w_frame52_local_ip.pcap, the packets sent from the local IP has the contents correct.
It is only after passing the public IP and at the network premises, that this failure is reported by the Session Border controller.
I am not able to understand why a correctly formed packet sent out is seen "corrupted" at the network end.
Any ideas really appreciated.
Thanks,
Badri.
Hello,
the conference mixer is supposed to send PUBLISH requests with
information about who is in the conference room, who is speaking at that
moment and so. That is not the job of the presence_conference module.
Eventually, that could be a pua_conference module that has to be
implemented. The presence_conference module was implemented as part of
GSoC and the PUA was supposed to be inside SEMS, but the other project
failed.
If you want to get it all working, you have to implement the PUA either
in your version of rtpproxy or part of kamailio as well. When all in
kamailio, you can see the pairs of modules such as presence_dialoginfo
and pua_dialoginfo. You already have presence_confgerece.
Cheers,
Daniel
On 28/05/15 14:28, Surendra Pullaiah wrote:
>
> Hello Daniel,
>
>
>
> I am implementing conference bridge using kamalio and
> RTP porxy without media mixing. Since for now we are ok with this.
> My question is as per the flow of RFC 4579, UA may send SUBSCRIBE to
> the conference URI with event as conference. In kamalio modules
> presence_conference where they mentioned such way that it will support
> the same what I am looking for. But I simulated using sipP by sending
> subscriber with conference event. I am not seeing any notify XML which
> will have the conference users list. One more doubt where it will pick
> the conference users, since I am not seeing any specific data base
> table for the same.
>
>
>
> Can you please clarify what exactly for
> presence_conference module.
>
>
>
> Note: I have loaded three modules presence, presence_xml,
> presence_conference, in route config I am calling handle_subscribe
> when SUBCRIBE method receives.
>
>
>
> Regards
>
> Surendra
>
> India: +91-8124625001
>
>
>
>
>
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--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
Hello,
I am trying to forward the register to multiple Asterisk using Kamailio.
The basic Idea is:
Softphone A---->Kamailio-----> Asterisk 1
------> Asterisk 2.
I follow Asipto howto http://lylix.net/kamailio, but the problem I see in
my case, is that registration IP in Asterisk is Kamailio´s IP. This causes
calls going from kamailio to asterisk and then back to kamailio. In my
scenario, I need that softphone A IP appeard in Asterisk realtime contact
IP.
The call path would be:
Softphone B -> kamailio -> Asterisk 1 or Asterisk 2 (depending on
dispatcher) -> Softphone A
Following Asipto how to the call goes Softphone A -> kamailio -> asterisk
-> kamailio Softphone B
Hope I explain myself clearly.
Thank you in advance.
HI!
I need help to configure RTPProxy.
I have working Kamailio on Centos 6.5, On LAN everything is working fine.
Now I want my other users to connect using 3G or any other network.
Please help me with that.
Statis Public IP (DSL Modem) ----- Proxy Box with 2 Lan cards ---
192.168.2.0 and 192.168.0.0 (Lan Side)
