[asterisk-bugs] [JIRA] (ASTERISK-23191) Cannot get local SDP on incoming calls (WebRTC)

[Jay Jideliov (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23191?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=214466#comment-214466 ] 

Jay Jideliov commented on ASTERISK-23191:
-----------------------------------------

Reliably Transmitting (NAT) to 1.1.1.1.1:22596:
INVITE sip:1061 at df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
Via: SIP/2.0/WS 2.2.2.2:5060;branch=z9hG4bK5d6845a4;rport
Max-Forwards: 70
From: "New User" <sip:1060 at 2.2.2.2>;tag=as07bcc9b2
To: <sip:1061 at df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:1060 at 2.2.2.2:5060;transport=WS>
Call-ID: 654bc8580334ada51c6cd2d75b83f46a at 162.209.101.19:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0-rc1
Date: Mon, 18 Nov 2013 11:59:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1083

v=0
o=root 327695973 327695973 IN IP4 2.2.2.2
s=Asterisk PBX 11.7.0-rc1
c=IN IP4 2.2.2.2
t=0 0
m=audio 16460 UDP/TLS/RTP/SAVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:32c9e6db46a817e61ddb57ff5662dcdd
a=ice-pwd:340e63ed4158b8dc6655a51968eb2392
a=candidate:Ha2d16513 1 UDP 2130706431 2.2.2.2 16460 typ host
a=candidate:Hab0006b 1 UDP 2130706431 10.176.0.107 16460 typ host
a=candidate:Hc0a8a808 1 UDP 2130706431 192.168.168.8 16460 typ host
a=candidate:Sa2d16513 1 UDP 1694498815 2.2.2.2 16460 typ srflx
a=candidate:Ha2d16513 2 UDP 2130706430 2.2.2.2 16461 typ host
a=candidate:Hab0006b 2 UDP 2130706430 10.176.0.107 16461 typ host
a=candidate:Hc0a8a808 2 UDP 2130706430 192.168.168.8 16461 typ host
a=candidate:Sa2d16513 2 UDP 1694498814 2.2.2.2 16460 typ srflx
a=connection:new
a=setup:active
a=fingerprint:SHA-1 12:D1:5C:DF:16:6A:54:2D:8E:B0:51:3B:6E:CC:F7:4E:2B:0B:08:2B
a=sendrecv
                
> Cannot get local SDP on incoming calls (WebRTC)
> -----------------------------------------------
>
>                 Key: ASTERISK-23191
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23191
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_http_websocket
>    Affects Versions: 11.7.0
>            Reporter: Jay Jideliov
>
> This is an issue created as a follow-up to this comment (https://issues.asterisk.org/jira/browse/ASTERISK-21930?focusedCommentId=211742&page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-211742) as well as due to us having the same issue when trying to receive incoming calls via WebRTC.
> Asterisk 11.7 + SIPML5 + WSS, both Firefox and Chrome.
> Devices register fine, and are able to make outgoing calls. However, the issue arises on incoming calls, when all we get is the error (cannot get local sdp) after which the line disconnects.

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