[Spce-user] OnSip Sip.Js WebRTC Libraries and SipWise

[Tarak Patel]

Hi Andrew,

After applying those changes when I do ngcpcfg apply I got following error.

Not executing command for kamailio-proxy: invalid configuration file!
-e
 0(2674) : <core> [cfg.y:3412]: yyerror_at(): parse error in config file /etc/kamailio/proxy/proxy.cfg, line 8773, column 2: syntax error
 0(2674) : <core> [cfg.y:3412]: yyerror_at(): parse error in config file /etc/kamailio/proxy/proxy.cfg, line 8773, column 2: bad command
 0(2674) : <core> [cfg.y:3412]: yyerror_at(): parse error in config file /etc/kamailio/proxy/proxy.cfg, line 8773, column 2: bad command
 0(2674) : <core> [cfg.y:3409]: yyerror_at(): parse error in config file /etc/kamailio/proxy/proxy.cfg, line 8775, column 9-10:
ERROR: bad config file (4 errors)

Thanks,
Tarak Patel

-----Original Message-----
From: Andrew Pogrebennyk [mailto:apogrebennyk at sipwise.com] 
Sent: Wednesday, March 18, 2015 11:34 AM
To: Tarak Patel; spce-user at lists.sipwise.com
Subject: Re: [Spce-user] OnSip Sip.Js WebRTC Libraries and SipWise

On 03/18/2015 04:03 PM, Tarak Patel wrote:
> Actually I am not getting any media at all. Please find the attached logs.

I have not tested the DTLS-SRTP recently. The rtpengine support for UDP/TLS/RTP/SAVPF profiles is in place, but the kamailio config is still using the obsolete SDES encryption.
We are thinking about switching completely to UDP/TLS/RTP/SAVPF in one of the nearest releases.

For now you can try the following:
1) set preferences srtp_transcoding and rtcp_feedback to Transparent on the domain level (and on subscribers if you have ever changed it there from the default..)

2) Change your
/etc/ngcp-config/templates/etc/kamailio/proxy/proxy.cfg.customtt.tt2 as below (create this file if it doesn't exist according to
https://www.sipwise.com/doc/mr3.7.2/spce/ar01s11.html#_tt2_and_customtt_tt2_files)
and execute ngcpcfg apply:

                setbflag(FLB_ICE_CALLER_STRIP);
        }

-       # webrtc endpoint automatic detection
-       $(avp(s:mline)[*]) = $null;
-       if(sdp_get_line_startswith("$avp(s:mline)","m=") &&
-          $avp(s:mline) =~ "SAVPF")
-       {
-               resetbflag(FLB_SAVP_CALLER_RTP);
-               setbflag(FLB_SAVP_CALLER_SRTP);
-       }
-       else if(
+       if(
           $var(from_local_endpoint) == 1 &&
           # if from voicemail or iaxmodem - transcode to RTP
           !($avp(s:from_faxserver) == 1 && $var(sendfax) != 1) @@ -7949,15 +7941,7 @@ route[ROUTE_BRANCH_ACC_RTP]
                setbflag(FLB_SAVP_CALLER_RTP);
        }

-       # webrtc endpoint automatic detection
-       $(avp(s:mline)[*]) = $null;
-       if(sdp_get_line_startswith("$avp(s:mline)","m=") &&
-          $avp(s:mline) =~ "SAVPF")
-       {
-               resetbflag(FLB_AVPF_CALLER_AVP);
-               setbflag(FLB_AVPF_CALLER_AVPF);
-       }
-       else if($avp(s:first_caller_rtcp_feedback) == "force_avpf")
+       if($avp(s:first_caller_rtcp_feedback) == "force_avpf")
        {
                add_rr_param(";avpf_caller=force_avpf");
                setbflag(FLB_AVPF_CALLER_AVPF); @@ -7988,13 +7972,7 @@ route[ROUTE_BRANCH_ACC_RTP]
                setbflag(FLB_ICE_CALLEE_STRIP);
        }

-       # webrtc endpoint automatic detection
-       if($(ru{uri.param,transport}) == "ws")
-       {
-               resetbflag(FLB_SAVP_CALLEE_RTP);
-               setbflag(FLB_SAVP_CALLEE_SRTP);
-       }
-       else if($var(to_local_endpoint) == 1 && $var(callee_is_hwfax) != 1)
+       if($var(to_local_endpoint) == 1 && $var(callee_is_hwfax) != 1)
        # don't transcode to RTP call to hwfax because this overwrites "UDPTL" to "RTP/AVP"
        {
                add_rr_param(";savp_callee=force_rtp");


Basically, we need to remove the conditions for webrtc endpoint automatic detection.

Andrew