VoIP Mailing List Archives

[ohjelmistoarkkitehti a...]

Hi,

Is there anything I can do with this problem? Re-installing Asterisk does not solve this and the problem still persists. Or is there any other logs or configurations I can provide to help figure out why Asterisk is removing lines from the sdp?


Any ideas would be greatly appreciated! I also tried removing everything under /etc/asterisk/ and make samples to restore any errors I could have had in my configurations, then restoring my minimal configuration: asterisk.conf, extconfig.conf, extensions.conf, res_mysql.conf and sip.conf. This did not help.


(in case this message comes double, I just canceled posting of previous similar one as it was too big)


cheers,
Olli 

[EWieling at nyigc.com]

Asterisk is not a SIP Proxy.   It is a B2BUA and will *always* replace the SDP with its own.

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Olli Heiskanen
Sent: Thursday, October 02, 2014 9:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients



Hi,


Is there anything I can do with this problem? Re-installing Asterisk does not solve this and the problem still persists. Or is there any other logs or configurations I can provide to help figure out why Asterisk is removing lines from the sdp?



Any ideas would be greatly appreciated! I also tried removing everything under /etc/asterisk/ and make samples to restore any errors I could have had in my configurations, then restoring my minimal configuration: asterisk.conf, extconfig.conf, extensions.conf, res_mysql.conf and sip.conf. This did not help.



(in case this message comes double, I just canceled posting of previous similar one as it was too big)



cheers,

Olli

[ohjelmistoarkkitehti a...]

Hi,

Thanks Eric for your reply, yes I know Asterisk replaces the sdp, however it should create ice lines when calling to a webrtc client, which it is currently not doing. 


To recap my problem (check previous messages for details); I have 2 webrtc clients (sip.js on chrome) with realtime information that appears to be correct. When calling from A to B, INVITE coming to Asterisk contains correct sdp, but when the INVITE leaves Asterisk, the sdp lacks ice lines. 


cheers,
Olli


2014-10-02 18:13 GMT+03:00 Eric Wieling <EWieling@nyigc.com (EWieling@nyigc.com)>:

[mjordan at digium.com]

On Thu, Oct 2, 2014 at 10:18 AM, Olli Heiskanen
<ohjelmistoarkkitehti@gmail.com> wrote:

[ohjelmistoarkkitehti a...]

Hi,

Thanks Matthew for trying to reproduce the problem, I appreciate your efforts very much.


There must be something off in my setup in one way or another. I could just discard this server and build a new one, but I think it's not good practice to leave a problem unsolved, so I'll continue trying to figure this out. One thing I noticed - don't know if it's relevant or not - due to a repo mismatch, I had problems with updating libgdiplus and libgdiplus-devel package, had to disable a repo and reinstall those and my mono installation (which is making me lose my hair).


Is there a way to debug Asterisk itself? Or find the code that parses the outbound sdp? I figured there must be an if statement or more that determines whether or not to parse the ice lines into the sdp body. Finding that/those statements that produce the kind of sdp I'm seeing Asterisk send out, might tell something about what's wrong with my setup. As my c is not exactly fluent I wasn't sure which code files to search, can you guys help out with that? 


cheers,
Olli






2014-10-03 11:31 GMT+03:00 Matthew Jordan <mjordan@digium.com (mjordan@digium.com)>:

[jcolp at digium.com]

Olli Heiskanen wrote:

[ohjelmistoarkkitehti a...]

Hi Joshua,

Excellent! I didn't even remember to consider newer versions of asterisk as 11.11 was the latest one when I started building on. I have had libuuid and libuuid-devel installed the whole time, but perhaps 11.11 just did not see it there. I just installed 11.13 and it works perfectly.


Thank you sir, I will raise a drink for you next time I'm out.


cheers,
Olli


2014-10-07 16:55 GMT+03:00 Joshua Colp <jcolp@digium.com (jcolp@digium.com)>: