no ringback tone (sip.js+freeswitch)

[hide]

hi,guys

          I use sip.js + freeswitch  and  make a call from chrome(extension 1000) to  x-lite(extension 1001) that both register on freeswitch.

The call is ok. But I met an issue that when I call from chrome to x-lite, I cant hear ringback tone from webbrowser.

         The sip 183 seems ok.

         anyone met this issue and how to fix it.

SIP/2.0 183 Session Progress

Via: SIP/2.0/WSS 5klp875mocgo.invalid;branch=z9hG4bK4649758;received=xxx.xx.xxx.xx;rport=56526

From: "1004" <sip:10...@aa.a.aa.aa>;tag=di6ppddffv

To: <sip:10...@aa.a.aa.aa>;tag=Q428y52ZBe9QK

Call-ID: s04jdkq0fd23icca36bj

CSeq: 6061 INVITE

Contact: <sip:10...@aa.a.aa.aa:5060;transport=udp>

User-Agent: FreeSWITCH-mod_sofia/1.6.13~64bit

Accept: application/sdp

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, path, replaces

Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 875

Remote-Party-ID: "1001" <sip:10...@aa.a.aa.aa>;party=calling;privacy=off;screen=no

v=0

o=FreeSWITCH 1483498276 1483498277 IN IP4 aa.a.aa.aa

s=FreeSWITCH

c=IN IP4 aa.a.aa.aa

t=0 0

a=msid-semantic: WMS Ro9AIF1IdC1wE11cydaJp03jiHP9GNzn

m=audio 22416 UDP/TLS/RTP/SAVPF 111 126

a=rtpmap:111 opus/48000/2

a=fmtp:111 useinbandfec=1; minptime=10

a=rtpmap:126 telephone-event/8000

a=ptime:20

a=fingerprint:sha-256 04:6B:BE:0C:CC:D1:9B:39:A2:A4:88:98:D0:34:45:BE:76:01:A7:6E:1F:5A:A3:41:F4:72:76:51:C6:CA:88:7D

a=setup:active

a=rtcp-mux

a=rtcp:22416 IN IP4 aa.a.aa.aa

a=ice-ufrag:pGQExKNGhsFN641X

a=ice-pwd:tGoodnlceGGWSc38IJD5jgBr

a=candidate:4365361948 1 udp 659136 aa.a.aa.aa 22416 typ host generation 0

a=end-of-candidates

a=ssrc:2154856963 cname:1zOA4G8790uDSmGg

a=ssrc:2154856963 msid:Ro9AIF1IdC1wE11cydaJp03jiHP9GNzn a0

a=ssrc:2154856963 mslabel:Ro9AIF1IdC1wE11cydaJp03jiHP9GNzn

a=ssrc:2154856963 label:Ro9AIF1IdC1wE11cydaJp03jiHP9GNzna0

          

          

[James Criscuolo]

Hi hide,

  This is likely an issue with your freeswitch configuration. I believe you need to set up early negotiation, so that there is a call up between freeswitch and your browser, before the other leg is set up. This also necessitates that the other leg is sending ringback, which is not guaranteed. For our webrtc calls at OnSIP, we manually play a tone on the browser side to simulate the experience.

James

[maricat...@gmail.com]

James, I have the same issue. I have tried several things in fs configuration:

<action application="pre_answer"/>
<action application="ring_ready"/>
<action application="set" data="ringback=%(2000, 4000, 440.0, 480.0)"/>

etc.

None of these seem to work for sipjs. Do you know if it ignores 180, 183 SIP packets or the incoming media?

Also, can you provide the setup for doing it locally in the browser?



Best regards.