[SR-Users] WebRTC error code 488 in iOS application

[Denys Pozniak]

Hello!

Maybe it will be useful for somebody, I applied next configuration and iOS
starts working.

....
modparam("rtpengine", "write_sdp_pv", "$avp(sdp)")
....
rtpengine_manage("trust-address replace-origin replace-session-connection
rtcp-mux-offer rtcp-mux-accept media-handover ICE=force RTP/SAVPF");
set_body("$avp(sdp)", "application/sdp");
....
$avp(sdp) = $(avp(sdp){re.subst,/t=0 0/t=0 0\r\na=group:BUNDLE audio/g});
$avp(sdp) =
$(avp(sdp){re.subst,/a=setup:actpass/a=setup:actpass\r\na=mid:audio/g});
set_body("$avp(sdp)", "application/sdp");

BR,
Denys




On 31 October 2017 at 16:57, Denys Pozniak <denys.pozniak at crazycall.com>
wrote:

> Hello!
>
> Probably this case is not connected Kamailio directly, but it can be
> somebody will point me in the correct direction :)
>
> We have VoIP solution: FreeSWITCH -> Kamailio + Rtpengine -> WebRTC Chrome.
> Everything works Ok except iOS application based on React Native WebRTC (
> https://github.com/oney/react-native-webrtc) + JsSIP.
> It answers error 488 on incoming INVITE (wSDP):
>
> INVITE SDP to iOS:
>
> *......*
> *v=0*
> *o=FreeSWITCH 1508701860 1508701861 IN IP4 52.52.52.52*
> *s=FreeSWITCH*
> *c=IN IP4 52.52.52.52*
> *t=0 0*
> *m=audio 30028 RTP/SAVPF 0 8 102*
> *a=rtpmap:0 PCMU/8000*
> *a=rtpmap:8 PCMA/8000*
> *a=rtpmap:102 opus/48000/2*
> *a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000;
> ptime=20; minptime=10; maxptime=40*
> *a=ptime:20*
> *a=sendrecv*
> *a=rtcp:30028*
> *a=rtcp-mux*
> *a=setup:actpass*
> *a=fingerprint:sha-1
> 97:1A:E9:FE:D6:65:98:E6:88:EE:D5:9F:20:A3:19:F8:86:E7:E0:E3*
> *a=ice-ufrag:9QBO8qg2*
> *a=ice-pwd:PPGPo35xGBAs6RnTdMYVM9I8fx*
> *a=candidate:KXENzrffZpQvwxdV 1 UDP 2130706431 52.52.52.52 30028 typ host*
>
>
> iOS console log:
>
>
> *11:55:23.670 browser.js:133 JsSIP:WebSocketInterface send() +10ms*
> *11:55:23.670 browser.js:133 JsSIP:RTCSession session progress +5ms*
> *11:55:23.671 browser.js:133 JsSIP:RTCSession emit "progress" +1ms*
> *11:55:23.671 webrtc.js:108 WebRTC.on progress:  local*
> *11:55:23.679 browser.js:133 JsSIP:RTCSession emit "peerconnection" +8ms*
> *11:55:23.680 webrtc.js:89 WebRTC.on peerconnection*
> *11:55:23.680 browser.js:133 JsSIP:RTCSession emit "sdp" +1ms*
> *11:55:23.689 browser.js:133 JsSIP:Transport send() +20ms**11:55:23.689
> browser.js:133 JsSIP:Transport sending message:*
> *SIP/2.0 488 Not Acceptable Here*
> *11:55:23.701 debug.js:115 JsSIP:ERROR:RTCSession emit
> "peerconnection:setremotedescriptionfailed" [error:Error: Failed to set
> remote offer sdp: Failed to create channels.*
>
>
>
> I found similar issue on thier git https://github.com/oney/
> react-native-webrtc/issues/293 and main solution is to struct SDP in way
> like:
>
>
>
> *a=group:BUNDLE audiom=audio PORT RTP/SAVPF N M Ka=mid:audio*
>
>
> How to modify SDP on Kamailio side (after Rtpengine modification) like
> below?
>
>
> *v=0*
> *o=FreeSWITCH 1508701860 1508701861 IN IP4 52.52.52.52*
> *s=FreeSWITCH*
> *c=IN IP4 52.52.52.52*
> *t=0 0*
> *a=group:BUNDLE audio*
> *m=audio 30028 RTP/SAVPF 0 8 102*
> *a=rtpmap:0 PCMU/8000*
> *a=rtpmap:8 PCMA/8000*
> *a=rtpmap:102 opus/48000/2*
> *a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000;
> ptime=20; minptime=10; maxptime=40*
> *a=ptime:20*
> *a=sendrecv*
> *a=rtcp:30028*
> *a=rtcp-mux*
> *a=setup:actpass*
> *a=mid:audio*
> *a=fingerprint:sha-1
> 97:1A:E9:FE:D6:65:98:E6:88:EE:D5:9F:20:A3:19:F8:86:E7:E0:E3*
> *a=ice-ufrag:9QBO8qg2*
> *a=ice-pwd:PPGPo35xGBAs6RnTdMYVM9I8fx*
> *a=candidate:KXENzrffZpQvwxdV 1 UDP 2130706431 52.52.52.52 30028 typ host*
>
>
>
> BR,
> Denys
>
>
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