[asterisk-bugs] [JIRA] (ASTERISK-24557) WebRTC call returns error "Failed to get local SDP"

Matt Jordan (JIRA) noreply at issues.asterisk.org
Tue Dec 23 13:59:35 CST 2014


    [ https://issues.asterisk.org/jira/browse/ASTERISK-24557?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224212#comment-224212 ] 

Matt Jordan commented on ASTERISK-24557:
----------------------------------------

Looking at your log file, the error message is being returned from whatever client Asterisk is communicating with:

{noformat}
<--- SIP read from WS:192.168.34.145:2562 --->
SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 192.168.34.199:5060;rport=5060;branch=z9hG4bK589fe139
From: "156"<sip:156 at 192.168.34.199>;tag=as451e8620
To: <sip:157 at df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=8W672ie5gHvCPbzyA7VQ
Call-ID: 597126967b118c560a03e3eb6393f8da at 192.168.34.199:5060
CSeq: 102 INVITE
Content-Length: 0
Reason: SIP; cause=603; text="Failed to get local SDP"

<------------->
{noformat}

You'll need to investigate it to determine what it doesn't like about the INVITE request Asterisk sent to it.

{noformat}
Reliably Transmitting (NAT) to 192.168.34.145:2562:
INVITE sip:157 at df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0

Via: SIP/2.0/WS 192.168.34.199:5060;branch=z9hG4bK589fe139;rport

Max-Forwards: 70

From: "156" <sip:156 at 192.168.34.199>;tag=as451e8620

To: <sip:157 at df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>

Contact: <sip:156 at 192.168.34.199:5060;transport=WS>

Call-ID: 597126967b118c560a03e3eb6393f8da at 192.168.34.199:5060

CSeq: 102 INVITE

User-Agent: Asterisk PBX SVN-branch-11-r429539

Date: Mon, 22 Dec 2014 07:40:34 GMT

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

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 409



v=0

o=root 1170437591 1170437591 IN IP4 192.168.34.199

s=Asterisk PBX SVN-branch-11-r429539

c=IN IP4 192.168.34.199

t=0 0

m=audio 12760 UDP/TLS/RTP/SAVPF 0 3 8

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:8 PCMA/8000

a=ptime:20

a=connection:new

a=setup:actpass

a=fingerprint:SHA-256 62:8E:8B:50:E7:8D:D6:62:DE:3B:9E:D8:E5:B8:49:04:23:E8:34:AF:C5:93:94:FF:DD:4E:9A:32:8D:C7:B8:82

a=sendrecv

{noformat}

That being said, it appears as if you've modified Asterisk. Asterisk does not include the request line modifier "rtcweb-breaker" anywhere.
{{INVITE sip:157 at df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0}}

The project does not provide source for modified versions of Asterisk.


> WebRTC call returns error "Failed to get local SDP"
> ---------------------------------------------------
>
>                 Key: ASTERISK-24557
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24557
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 11.9.0, 11.14.1
>         Environment: centos6.6, debian7.7.0 (i686 and x64)
>            Reporter: Osaulenko Alexander
>            Assignee: Matt Jordan
>         Attachments: backtrace_20141222.txt, backtrace(SDP fail).txt, backtrace.txt
>
>
> We use Asterisk of different versions 11.9-11.14.1 with default settings for WebRTC and Openssl version: OpenSSL 1.0.1e 11 Feb 2013. It works but craches on:
> d1_both.c(278): OpenSSL internal error, assertion failed: s->init_num == (int)s->d1->w_msg_hdr.msg_len + DTLS1_HM_HEADER_LENGTH
> We tried update openssl to OpenSSL 1.0.1j and then compiled Asterisk.
> When we try to call using WebRTC we have error "Failed to get local SDP"



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