*RESOLVED*Call to Chrome fails:603 Failed to get local SDP

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*RESOLVED*Call to Chrome fails:603 Failed to get local SDP

Postby inno2k » Tue Mar 25, 2014 4:15 pm

Greetings all,

I have been struggling with this for a while now, so I am hoping someone can give me some advice.

I am trying to setup a WEBRTC capable Asterisk server that will allow calls via Chrome only within our Internal network.

I've got it setup and working where calls work from Browser -> Softphone (X-Lite) and Softphone -> Softphone.

The problem is when we try to make a call to the browser, upon accepting permission to use the MIC, the browser produces a "Call Rejected" message.

On the Asterisk logging it shows: Failed to get local SDP

Can anyone point me in the right direction? Thanks!

Asterisk Debugging shows this:

[Mar 25 17:09:59] DEBUG[29096] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing
[Mar 25 17:09:59] DEBUG[29096] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/WS 10.168.100.160:5060;rport=5060;branch=z9hG4bK51dcf9c7
[Mar 25 17:09:59] DEBUG[29096] chan_sip.c: Header 2 [ 51]: From: "Tim"<sip:2002@10.168.100.160>;tag=as76684801
[Mar 25 17:09:59] DEBUG[29096] chan_sip.c: Header 3 [ 92]: To: <sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=Bw5FZFL1H13kz43AzG3P
[Mar 25 17:09:59] DEBUG[29096] chan_sip.c: Header 4 [ 53]: Contact: <sip:josh@df7jal23ls0d.invalid;transport=ws>
[Mar 25 17:09:59] DEBUG[29096] chan_sip.c: Header 5 [ 61]: Call-ID: 673e951e215749e83fb8612957b963d4@10.168.100.160:5060
[Mar 25 17:09:59] DEBUG[29096] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE
[Mar 25 17:09:59] DEBUG[29096] chan_sip.c: Header 7 [ 17]: Content-Length: 0
[Mar 25 17:09:59] DEBUG[29096] chan_sip.c: Header 8 [ 79]: Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
[Mar 25 17:09:59] VERBOSE[29096] chan_sip.c: --- (9 headers 0 lines) ---
[Mar 25 17:09:59] DEBUG[29096] chan_sip.c: = Looking for Call ID: 673e951e215749e83fb8612957b963d4@10.168.100.160:5060 (Checking To) --From tag as76684801 --To-tag Bw5FZFL1H13kz43AzG3P
[Mar 25 17:09:59] DEBUG[29096][C-00000003] chan_sip.c: SIP response 180 to standard invite
[Mar 25 17:09:59] DEBUG[29096][C-00000003] chan_sip.c: build_route: Contact hop: <sip:josh@df7jal23ls0d.invalid;transport=ws>
[Mar 25 17:09:59] VERBOSE[29096][C-00000003] chan_sip.c: list_route: hop: <sip:josh@df7jal23ls0d.invalid;transport=ws>
[Mar 25 17:09:59] DEBUG[29051] devicestate.c: No provider found, checking channel drivers for SIP - josh
[Mar 25 17:09:59] DEBUG[29051] chan_sip.c: Checking device state for peer josh
[Mar 25 17:09:59] DEBUG[29051] devicestate.c: Changing state for SIP/josh - state 1 (Not in use)
[Mar 25 17:09:59] DEBUG[29051] devicestate.c: device 'SIP/josh' state '1'
[Mar 25 17:09:59] DEBUG[29086] app_queue.c: Device 'SIP/josh' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 25 17:09:59] VERBOSE[29190][C-00000003] app_dial.c: -- SIP/josh-00000007 is ringing
[Mar 25 17:09:59] DEBUG[29190][C-00000003] rtp_engine.c: Setting early bridge SDP of 'SIP/tim-00000006' with that of 'SIP/josh-00000007'
[Mar 25 17:09:59] VERBOSE[29190][C-00000003] chan_sip.c:
ÿ<--- Transmitting (no NAT) to 10.168.107.7:5060 --->
ÿSIP/2.0 180 Ringing
ÿVia: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKaPLzsqU7ZgJmWThgGmvpIWZLGHD6xpoP;rport;received=10.168.107.7
ÿFrom: "tim"<sip:tim@10.168.100.160>;tag=TVH5FAXeAvbD76It71Ny
ÿTo: <sip:2001@10.168.100.160>;tag=as2fad071f
ÿCall-ID: 0fdd5138-08a6-972a-c5fa-ce4e125da644
ÿCSeq: 14184 INVITE
ÿServer: Asterisk PBX 11.7.0
ÿAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
ÿSupported: replaces, timer
ÿContact: <sip:2001@10.168.100.160:5060;transport=WS>
ÿContent-Length: 0
ÿ
ÿ
ÿ<------------>
[Mar 25 17:09:59] DEBUG[29190][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 180' onto WS socket destined for 10.168.107.7:5060

ÿ<------------->
[Mar 25 17:10:01] DEBUG[29076] chan_sip.c: Header 0 [ 0]:
[Mar 25 17:10:01] VERBOSE[29096] chan_sip.c:
ÿ<--- SIP read from WS:10.168.107.7:64887 --->
ÿSIP/2.0 603 Failed to get local SDP
ÿVia: SIP/2.0/WS 10.168.100.160:5060;rport=5060;branch=z9hG4bK51dcf9c7
ÿFrom: "Tim"<sip:2002@10.168.100.160>;tag=as76684801
ÿTo: <sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=Bw5FZFL1H13kz43AzG3P
ÿCall-ID: 673e951e215749e83fb8612957b963d4@10.168.100.160:5060
ÿCSeq: 102 INVITE
ÿContent-Length: 0
ÿReason: SIP; cause=603; text="Failed to get local SDP"
ÿ
ÿ<------------->
[Mar 25 17:10:01] DEBUG[29096] chan_sip.c: Header 0 [ 35]: SIP/2.0 603 Failed to get local SDP
[Mar 25 17:10:01] DEBUG[29096] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/WS 10.168.100.160:5060;rport=5060;branch=z9hG4bK51dcf9c7
[Mar 25 17:10:01] DEBUG[29096] chan_sip.c: Header 2 [ 51]: From: "Tim"<sip:2002@10.168.100.160>;tag=as76684801
[Mar 25 17:10:01] DEBUG[29096] chan_sip.c: Header 3 [ 92]: To: <sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=Bw5FZFL1H13kz43AzG3P
[Mar 25 17:10:01] DEBUG[29096] chan_sip.c: Header 4 [ 61]: Call-ID: 673e951e215749e83fb8612957b963d4@10.168.100.160:5060
[Mar 25 17:10:01] DEBUG[29096] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE
[Mar 25 17:10:01] DEBUG[29096] chan_sip.c: Header 6 [ 17]: Content-Length: 0
[Mar 25 17:10:01] DEBUG[29096] chan_sip.c: Header 7 [ 54]: Reason: SIP; cause=603; text="Failed to get local SDP"
[Mar 25 17:10:01] VERBOSE[29096] chan_sip.c: --- (8 headers 0 lines) ---
[Mar 25 17:10:01] DEBUG[29096] chan_sip.c: = Looking for Call ID: 673e951e215749e83fb8612957b963d4@10.168.100.160:5060 (Checking To) --From tag as76684801 --To-tag Bw5FZFL1H13kz43AzG3P
[Mar 25 17:10:01] VERBOSE[29096][C-00000003] chan_sip.c: -- Got SIP response 603 "Failed to get local SDP" back from 10.168.107.7:64887
[Mar 25 17:10:01] DEBUG[29096][C-00000003] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f92cc0137a8'
[Mar 25 17:10:01] DEBUG[29096][C-00000003] chan_sip.c: Strict routing enforced for session 673e951e215749e83fb8612957b963d4@10.168.100.160:5060
[Mar 25 17:10:01] VERBOSE[29096][C-00000003] chan_sip.c: set_destination: Parsing <sip:josh@df7jal23ls0d.invalid;transport=ws> for address/port to send to
[Mar 25 17:10:01] VERBOSE[29096][C-00000003] chan_sip.c: set_destination: URI is for WebSocket, we can't set destination
[Mar 25 17:10:01] VERBOSE[29096][C-00000003] chan_sip.c: Transmitting (NAT) to 10.168.107.7:64887:
ÿACK sip:josh@df7jal23ls0d.invalid;transport=ws SIP/2.0
ÿVia: SIP/2.0/WS 10.168.100.160:5060;branch=z9hG4bK51dcf9c7;rport
ÿMax-Forwards: 70
ÿFrom: "Tim" <sip:2002@10.168.100.160>;tag=as76684801
ÿTo: <sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=Bw5FZFL1H13kz43AzG3P
ÿContact: <sip:2002@10.168.100.160:5060;transport=WS>
ÿCall-ID: 673e951e215749e83fb8612957b963d4@10.168.100.160:5060
ÿCSeq: 102 ACK
ÿUser-Agent: Asterisk PBX 11.7.0
ÿContent-Length: 0
ÿ
ÿ
ÿ---
[Mar 25 17:10:01] DEBUG[29096][C-00000003] chan_sip.c: Trying to put 'ACK sip:jos' onto WS socket destined for 10.168.107.7:64887
[Mar 25 17:10:01] DEBUG[29096][C-00000003] chan_sip.c: Setting SIP_ALREADYGONE on dialog 673e951e215749e83fb8612957b963d4@10.168.100.160:5060
[Mar 25 17:10:01] VERBOSE[29190][C-00000003] app_dial.c: -- SIP/josh-00000007 is busy
[Mar 25 17:10:01] DEBUG[29190][C-00000003] channel.c: Hanging up channel 'SIP/josh-00000007'
[Mar 25 17:10:01] DEBUG[29190][C-00000003] chan_sip.c: Hangup call SIP/josh-00000007, SIP callid 673e951e215749e83fb8612957b963d4@10.168.100.160:5060
[Mar 25 17:10:01] DEBUG[29190][C-00000003] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f92cc0137a8'
[Mar 25 17:10:01] DEBUG[29051] devicestate.c: No provider found, checking channel drivers for SIP - josh
[Mar 25 17:10:01] DEBUG[29051] chan_sip.c: Checking device state for peer josh
[Mar 25 17:10:01] DEBUG[29051] devicestate.c: Changing state for SIP/josh - state 1 (Not in use)
[Mar 25 17:10:01] DEBUG[29051] devicestate.c: device 'SIP/josh' state '1'
[Mar 25 17:10:01] DEBUG[29086] app_queue.c: Device 'SIP/josh' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 25 17:10:01] VERBOSE[29190][C-00000003] app_dial.c: == Everyone is busy/congested at this time (1:1/0/0)
[Mar 25 17:10:01] DEBUG[29190][C-00000003] app_dial.c: Exiting with DIALSTATUS=BUSY.
[Mar 25 17:10:01] VERBOSE[29190][C-00000003] pbx.c: -- Auto fallthrough, channel 'SIP/tim-00000006' status is 'BUSY'
[Mar 25 17:10:01] VERBOSE[29190][C-00000003] chan_sip.c:
Last edited by inno2k on Wed Mar 26, 2014 5:13 pm, edited 1 time in total.
inno2k
Newsterisk
 
Posts: 10
Joined: Tue Mar 25, 2014 4:05 pm

Re: Incoming call to Chrome fails : 603 Failed to get local SDP

Postby navaismo » Tue Mar 25, 2014 6:37 pm

Show us the sip debug of asterisk without cli DEBUG and also the JS log from chrome.
navaismo
Salt of the Asterisk
 
Posts: 1610
Joined: Mon Dec 07, 2009 1:30 pm
Location: Mexico City, Mexico

Re: Incoming call to Chrome fails : 603 Failed to get local SDP

Postby inno2k » Tue Mar 25, 2014 7:53 pm

How's this?

Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
Asterisk 11.7.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 11.7.0 currently running on LaneAsterix (pid = 30119)

LaneAsterix*CLI>

<--- SIP read from WS:10.168.105.57:56932 --->
INVITE sip:2001@10.168.100.160 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKeAX5XxI0XuB9FRBeM9PANKFh5VN9PcBG;rport
From: "tim"<sip:tim@10.168.100.160>;tag=9RNQPG7ZIC9RmTFwT0b6
To: <sip:2001@10.168.100.160>
Contact: "tim"<sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=tim;ha1=41d8cc07daad5ff131cc9ae21713626b;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 7ca8707d-fb9e-14f2-31a1-fe98b77b170d
CSeq: 4637 INVITE
Content-Type: application/sdp
Content-Length: 2582
Route: <sip:10.168.100.160:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.03.26
Organization: Doubango Telecom

v=0
o=- 2849187087576387000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS uyB5eutn3ryDpNN1mlWjMYGRzK2GTs9tz9wx
m=audio 65236 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.56.1
a=rtcp:65236 IN IP4 192.168.56.1
a=candidate:2999745851 1 udp 2113937151 192.168.56.1 65236 typ host generation 0
a=candidate:2999745851 2 udp 2113937151 192.168.56.1 65236 typ host generation 0
a=candidate:1795485746 1 udp 2113937151 10.168.105.57 65237 typ host generation 0
a=candidate:1795485746 2 udp 2113937151 10.168.105.57 65237 typ host generation 0
a=candidate:828646434 1 udp 2113937151 192.168.220.1 65238 typ host generation 0
a=candidate:828646434 2 udp 2113937151 192.168.220.1 65238 typ host generation 0
a=candidate:1448336772 1 udp 2113937151 192.168.182.1 65239 typ host generation 0
a=candidate:1448336772 2 udp 2113937151 192.168.182.1 65239 typ host generation 0
a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=candidate:629607618 1 tcp 1509957375 10.168.105.57 0 typ host generation 0
a=candidate:629607618 2 tcp 1509957375 10.168.105.57 0 typ host generation 0
a=candidate:2145900754 1 tcp 1509957375 192.168.220.1 0 typ host generation 0
a=candidate:2145900754 2 tcp 1509957375 192.168.220.1 0 typ host generation 0
a=candidate:416293236 1 tcp 1509957375 192.168.182.1 0 typ host generation 0
a=candidate:416293236 2 tcp 1509957375 192.168.182.1 0 typ host generation 0
a=ice-ufrag:QtyI4J4u3ENY5UiS
a=ice-pwd:TtNGOl1CP2p8bgSUDczMf9wU
a=ice-options:google-ice
a=fingerprint:sha-256 F0:6D:1C:F5:93:4D:9F:EE:69:FA:34:42:AC:93:52:3D:26:13:87:CE:04:AF:D8:77:8B:83:5A:A2:57:2F:00:8F
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:TGeA4NeMbIhZTkSbcniStwdU5+RtzyDN80bw4jxL
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NqL89lbdDPbaEpPq7Gxch9A2UAhqzzs0nLXcePbG
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:2311922202 cname:H7NAk2NfUQSKoYcj
a=ssrc:2311922202 msid:uyB5eutn3ryDpNN1mlWjMYGRzK2GTs9tz9wx 4bddc6c6-20ba-4d36-ad8f-180375e1684c
a=ssrc:2311922202 mslabel:uyB5eutn3ryDpNN1mlWjMYGRzK2GTs9tz9wx
a=ssrc:2311922202 label:4bddc6c6-20ba-4d36-ad8f-180375e1684c
<------------->
--- (13 headers 51 lines) ---
Using INVITE request as basis request - 7ca8707d-fb9e-14f2-31a1-fe98b77b170d
Found peer 'tim' for 'tim' from 10.168.105.57:56932

<--- Reliably Transmitting (no NAT) to 10.168.105.57:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKeAX5XxI0XuB9FRBeM9PANKFh5VN9PcBG;rport;received=10.168.105.57
From: "tim"<sip:tim@10.168.100.160>;tag=9RNQPG7ZIC9RmTFwT0b6
To: <sip:2001@10.168.100.160>;tag=as06aacb65
Call-ID: 7ca8707d-fb9e-14f2-31a1-fe98b77b170d
CSeq: 4637 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2bd6509d"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '7ca8707d-fb9e-14f2-31a1-fe98b77b170d' in 32000 ms (Method: INVITE)

LaneAsterix*CLI>

<--- SIP read from WS:10.168.105.57:56932 --->
ACK sip:2001@10.168.100.160 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKeAX5XxI0XuB9FRBeM9PANKFh5VN9PcBG;rport
From: "tim"<sip:tim@10.168.100.160>;tag=9RNQPG7ZIC9RmTFwT0b6
To: <sip:2001@10.168.100.160>;tag=as06aacb65
Call-ID: 7ca8707d-fb9e-14f2-31a1-fe98b77b170d
CSeq: 4637 ACK
Content-Length: 0
Route: <sip:10.168.100.160:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70

<------------->
--- (9 headers 0 lines) ---

LaneAsterix*CLI>

<--- SIP read from WS:10.168.105.57:56932 --->
INVITE sip:2001@10.168.100.160 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKdFTgLHDLXx6N5AIaG7pDF1WbN5iJOJb2;rport
From: "tim"<sip:tim@10.168.100.160>;tag=9RNQPG7ZIC9RmTFwT0b6
To: <sip:2001@10.168.100.160>
Contact: "tim"<sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=tim;ha1=41d8cc07daad5ff131cc9ae21713626b;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 7ca8707d-fb9e-14f2-31a1-fe98b77b170d
CSeq: 4638 INVITE
Content-Type: application/sdp
Content-Length: 2582
Route: <sip:10.168.100.160:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="tim",realm="asterisk",nonce="2bd6509d",uri="sip:2001@10.168.100.160",response="68a0c58bbebffc327c2a125708beb4f2",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.03.26
Organization: Doubango Telecom

v=0
o=- 2849187087576387000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS uyB5eutn3ryDpNN1mlWjMYGRzK2GTs9tz9wx
m=audio 65236 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.56.1
a=rtcp:65236 IN IP4 192.168.56.1
a=candidate:2999745851 1 udp 2113937151 192.168.56.1 65236 typ host generation 0
a=candidate:2999745851 2 udp 2113937151 192.168.56.1 65236 typ host generation 0
a=candidate:1795485746 1 udp 2113937151 10.168.105.57 65237 typ host generation 0
a=candidate:1795485746 2 udp 2113937151 10.168.105.57 65237 typ host generation 0
a=candidate:828646434 1 udp 2113937151 192.168.220.1 65238 typ host generation 0
a=candidate:828646434 2 udp 2113937151 192.168.220.1 65238 typ host generation 0
a=candidate:1448336772 1 udp 2113937151 192.168.182.1 65239 typ host generation 0
a=candidate:1448336772 2 udp 2113937151 192.168.182.1 65239 typ host generation 0
a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=candidate:629607618 1 tcp 1509957375 10.168.105.57 0 typ host generation 0
a=candidate:629607618 2 tcp 1509957375 10.168.105.57 0 typ host generation 0
a=candidate:2145900754 1 tcp 1509957375 192.168.220.1 0 typ host generation 0
a=candidate:2145900754 2 tcp 1509957375 192.168.220.1 0 typ host generation 0
a=candidate:416293236 1 tcp 1509957375 192.168.182.1 0 typ host generation 0
a=candidate:416293236 2 tcp 1509957375 192.168.182.1 0 typ host generation 0
a=ice-ufrag:QtyI4J4u3ENY5UiS
a=ice-pwd:TtNGOl1CP2p8bgSUDczMf9wU
a=ice-options:google-ice
a=fingerprint:sha-256 F0:6D:1C:F5:93:4D:9F:EE:69:FA:34:42:AC:93:52:3D:26:13:87:CE:04:AF:D8:77:8B:83:5A:A2:57:2F:00:8F
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:TGeA4NeMbIhZTkSbcniStwdU5+RtzyDN80bw4jxL
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NqL89lbdDPbaEpPq7Gxch9A2UAhqzzs0nLXcePbG
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:2311922202 cname:H7NAk2NfUQSKoYcj
a=ssrc:2311922202 msid:uyB5eutn3ryDpNN1mlWjMYGRzK2GTs9tz9wx 4bddc6c6-20ba-4d36-ad8f-180375e1684c
a=ssrc:2311922202 mslabel:uyB5eutn3ryDpNN1mlWjMYGRzK2GTs9tz9wx
a=ssrc:2311922202 label:4bddc6c6-20ba-4d36-ad8f-180375e1684c
<------------->
--- (14 headers 51 lines) ---
Using INVITE request as basis request - 7ca8707d-fb9e-14f2-31a1-fe98b77b170d
Found peer 'tim' for 'tim' from 10.168.105.57:56932
== Using SIP RTP CoS mark 5
[Mar 26 12:29:19] NOTICE[32584][C-00000003]: chan_sip.c:10114 process_sdp: Received SAVPF profle in audio offer but AVPF is not enabled, enabling: audio 65236 RTP/SAVPF 111 103 104 0 8 106 105 13 126
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found unknown media description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.56.1:65236
Looking for 2001 in users (domain 10.168.100.160)
list_route: hop: <sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>

<--- Transmitting (no NAT) to 10.168.105.57:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKdFTgLHDLXx6N5AIaG7pDF1WbN5iJOJb2;rport;received=10.168.105.57
From: "tim"<sip:tim@10.168.100.160>;tag=9RNQPG7ZIC9RmTFwT0b6
To: <sip:2001@10.168.100.160>
Call-ID: 7ca8707d-fb9e-14f2-31a1-fe98b77b170d
CSeq: 4638 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2001@10.168.100.160:5060;transport=WS>
Content-Length: 0


<------------>

LaneAsterix*CLI>
 -- Executing [2001@users:1] Dial("SIP/tim-00000006", "SIP/josh,15") in new stack

LaneAsterix*CLI>
 == Using SIP RTP CoS mark 5

LaneAsterix*CLI>
[Mar 26 12:29:19] ERROR[32588][C-00000003]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Temporary failure in name resolution

LaneAsterix*CLI>
[Mar 26 12:29:19] WARNING[32588][C-00000003]: chan_sip.c:15881 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'

LaneAsterix*CLI>
[Mar 26 12:29:19] ERROR[32588][C-00000003]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported

LaneAsterix*CLI>
Audio is at 18284

LaneAsterix*CLI>
Adding codec 100003 (ulaw) to SDP

LaneAsterix*CLI>
Adding codec 100002 (gsm) to SDP

LaneAsterix*CLI>
Adding codec 100004 (alaw) to SDP

LaneAsterix*CLI>
Adding codec 100017 (testlaw) to SDP

LaneAsterix*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

LaneAsterix*CLI>
Reliably Transmitting (NAT) to 10.168.105.57:56930:
INVITE sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.168.100.160:5060;branch=z9hG4bK0c8ac5e2;rport
Max-Forwards: 70
From: "Tim" <sip:2002@10.168.100.160>;tag=as29fdee34
To: <sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:2002@10.168.100.160:5060;transport=WS>
Call-ID: 18093f87199f7e962457ab9910c9af5d@10.168.100.160:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 26 Mar 2014 16:29:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 514

v=0
o=root 1749118044 1749118044 IN IP4 10.168.100.160
s=Asterisk PBX 11.7.0
c=IN IP4 10.168.100.160
t=0 0
m=audio 18284 RTP/AVP 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=ptime:20
a=ice-ufrag:051ca7456520dc8a6169e06a30f669d1
a=ice-pwd:39c536aa4269f2d22fe08a994207cc51
a=candidate:Haa864a0 1 UDP 2130706431 10.168.100.160 18284 typ host
a=candidate:Haa864a0 2 UDP 2130706430 10.168.100.160 18285 typ host
a=sendrecv

---

LaneAsterix*CLI>
 -- Called SIP/josh

LaneAsterix*CLI>

<--- SIP read from WS:10.168.105.57:56930 --->
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 10.168.100.160:5060;rport=5060;branch=z9hG4bK0c8ac5e2
From: "Tim"<sip:2002@10.168.100.160>;tag=as29fdee34
To: <sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Call-ID: 18093f87199f7e962457ab9910c9af5d@10.168.100.160:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

LaneAsterix*CLI>

<--- SIP read from WS:10.168.105.57:56930 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 10.168.100.160:5060;rport=5060;branch=z9hG4bK0c8ac5e2
From: "Tim"<sip:2002@10.168.100.160>;tag=as29fdee34
To: <sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=O4rtluyBV9UhoiClOhTD
Contact: <sip:josh@df7jal23ls0d.invalid;transport=ws>
Call-ID: 18093f87199f7e962457ab9910c9af5d@10.168.100.160:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

<------------->

LaneAsterix*CLI>
--- (9 headers 0 lines) ---
list_route: hop: <sip:josh@df7jal23ls0d.invalid;transport=ws>

LaneAsterix*CLI>
 -- SIP/josh-00000007 is ringing

LaneAsterix*CLI>

<--- Transmitting (no NAT) to 10.168.105.57:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKdFTgLHDLXx6N5AIaG7pDF1WbN5iJOJb2;rport;received=10.168.105.57
From: "tim"<sip:tim@10.168.100.160>;tag=9RNQPG7ZIC9RmTFwT0b6
To: <sip:2001@10.168.100.160>;tag=as3ed2a4eb
Call-ID: 7ca8707d-fb9e-14f2-31a1-fe98b77b170d
CSeq: 4638 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2001@10.168.100.160:5060;transport=WS>
Content-Length: 0



LaneAsterix*CLI>

<--- SIP read from WS:10.168.105.57:56930 --->
SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 10.168.100.160:5060;rport=5060;branch=z9hG4bK0c8ac5e2
From: "Tim"<sip:2002@10.168.100.160>;tag=as29fdee34
To: <sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=O4rtluyBV9UhoiClOhTD
Call-ID: 18093f87199f7e962457ab9910c9af5d@10.168.100.160:5060
CSeq: 102 INVITE
Content-Length: 0
Reason: SIP; cause=603; text="Failed to get local SDP"

<------------->
--- (8 headers 0 lines) ---
-- Got SIP response 603 "Failed to get local SDP" back from 10.168.105.57:56930
set_destination: Parsing <sip:josh@df7jal23ls0d.invalid;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Transmitting (NAT) to 10.168.105.57:56930:
ACK sip:josh@df7jal23ls0d.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.168.100.160:5060;branch=z9hG4bK0c8ac5e2;rport
Max-Forwards: 70
From: "Tim" <sip:2002@10.168.100.160>;tag=as29fdee34
To: <sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=O4rtluyBV9UhoiClOhTD
Contact: <sip:2002@10.168.100.160:5060;transport=WS>
Call-ID: 18093f87199f7e962457ab9910c9af5d@10.168.100.160:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0
Content-Length: 0


---

LaneAsterix*CLI>
 -- SIP/josh-00000007 is busy

LaneAsterix*CLI>
 == Everyone is busy/congested at this time (1:1/0/0)

LaneAsterix*CLI>
 -- Auto fallthrough, channel 'SIP/tim-00000006' status is 'BUSY'

LaneAsterix*CLI>

<--- Reliably Transmitting (no NAT) to 10.168.105.57:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKdFTgLHDLXx6N5AIaG7pDF1WbN5iJOJb2;rport;received=10.168.105.57
From: "tim"<sip:tim@10.168.100.160>;tag=9RNQPG7ZIC9RmTFwT0b6
To: <sip:2001@10.168.100.160>;tag=as3ed2a4eb
Call-ID: 7ca8707d-fb9e-14f2-31a1-fe98b77b170d
CSeq: 4638 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0


<------------>

LaneAsterix*CLI>

<--- SIP read from WS:10.168.105.57:56930 --->
SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/WS 10.168.100.160:5060;rport=5060;branch=z9hG4bK0c8ac5e2
From: "Tim"<sip:2002@10.168.100.160>;tag=as29fdee34
To: <sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=O4rtluyBV9UhoiClOhTD
Call-ID: 18093f87199f7e962457ab9910c9af5d@10.168.100.160:5060
CSeq: 102 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
[Mar 26 12:29:21] WARNING[32583][C-00000003]: chan_sip.c:23919 handle_response: Remote host can't match request ACK to call '18093f87199f7e962457ab9910c9af5d@10.168.100.160:5060'. Giving up.

LaneAsterix*CLI>

<--- SIP read from WS:10.168.105.57:56932 --->
ACK sip:2001@10.168.100.160 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKdFTgLHDLXx6N5AIaG7pDF1WbN5iJOJb2;rport
From: "tim"<sip:tim@10.168.100.160>;tag=9RNQPG7ZIC9RmTFwT0b6
To: <sip:2001@10.168.100.160>;tag=as3ed2a4eb
Call-ID: 7ca8707d-fb9e-14f2-31a1-fe98b77b170d
CSeq: 4638 ACK
Content-Length: 0
Route: <sip:10.168.100.160:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70

<------------->
--- (9 headers 0 lines) ---

Last edited by inno2k on Wed Mar 26, 2014 10:32 am, edited 1 time in total.
inno2k
Newsterisk
 
Posts: 10
Joined: Tue Mar 25, 2014 4:05 pm

Re: Incoming call to Chrome fails : 603 Failed to get local SDP

Postby inno2k » Tue Mar 25, 2014 7:55 pm

sip_transport_ws_onmessage SIPml-api.js?svn=179:1
recv=INVITE sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.168.100.160:5060;rport;branch=z9hG4bK219f1ff4
From: "Josh"<sip:2001@10.168.100.160>;tag=as6beea29a
To: <sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:2001@10.168.100.160:5060;transport=WS>
Call-ID: 0da24b9c0306805c51ffe7461797a3ed@10.168.100.160:5060
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 512
Max-Forwards: 70
User-Agent: Asterisk PBX 11.7.0
Date: 26 Mar 2014 01:48:06 GMT;26
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

v=0
o=root 644042219 644042219 IN IP4 10.168.100.160
s=Asterisk PBX 11.7.0
c=IN IP4 10.168.100.160
t=0 0
m=audio 10848 RTP/AVP 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=ptime:20
a=ice-ufrag:5e7e0c712478a7ac53b905f62c50d3ea
a=ice-pwd:732a2d6c52ecf9dd48f47b08140ae21c
a=candidate:Haa864a0 1 UDP 2130706431 10.168.100.160 10848 typ host
a=candidate:Haa864a0 2 UDP 2130706430 10.168.100.160 10849 typ host
a=sendrecv
SIPml-api.js?svn=179:1
State machine: tsip_transac_ist_Started_2_Proceeding_X_INVITE SIPml-api.js?svn=179:1
SEND: SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 10.168.100.160:5060;rport=5060;branch=z9hG4bK219f1ff4
From: "Josh"<sip:2001@10.168.100.160>;tag=as6beea29a
To: <sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Call-ID: 0da24b9c0306805c51ffe7461797a3ed@10.168.100.160:5060
CSeq: 102 INVITE
Content-Length: 0

SIPml-api.js?svn=179:1
ICE servers:[] SIPml-api.js?svn=179:1
setRemoteDescription(offer)
v=0
o=root 644042219 644042219 IN IP4 10.168.100.160
s=Asterisk PBX 11.7.0
c=IN IP4 10.168.100.160
t=0 0
m=audio 10848 RTP/AVP 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=ptime:20
a=ice-ufrag:5e7e0c712478a7ac53b905f62c50d3ea
a=ice-pwd:732a2d6c52ecf9dd48f47b08140ae21c
a=candidate:Haa864a0 1 UDP 2130706431 10.168.100.160 10848 typ host
a=candidate:Haa864a0 2 UDP 2130706430 10.168.100.160 10849 typ host
a=sendrecv
SIPml-api.js?svn=179:1
State machine: s0000_Started_2_Ringing_X_iINVITE SIPml-api.js?svn=179:1
onSetRemoteDescriptionError SIPml-api.js?svn=179:1
SetRemoteDescription failed: Called with a SDP without crypto enabled. SIPml-api.js?svn=179:1
==stack event = m_permission_requested SIPml-api.js?svn=179:1
State machine: tsip_transac_ist_Proceeding_2_Proceeding_X_1xx SIPml-api.js?svn=179:1
SEND: SIP/2.0 180 Ringing
Via: SIP/2.0/WS 10.168.100.160:5060;rport=5060;branch=z9hG4bK219f1ff4
From: "Josh"<sip:2001@10.168.100.160>;tag=as6beea29a
To: <sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=7lSNF1ZRftqROULGsrJT
Contact: <sip:tim@df7jal23ls0d.invalid;transport=ws>
Call-ID: 0da24b9c0306805c51ffe7461797a3ed@10.168.100.160:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

SIPml-api.js?svn=179:1
==stack event = i_new_call SIPml-api.js?svn=179:1
onGetUserMediaSuccess SIPml-api.js?svn=179:1
createAnswer SIPml-api.js?svn=179:1
onCreateSdpError SIPml-api.js?svn=179:1
CreateAnswer can't be called before SetRemoteDescription. SIPml-api.js?svn=179:1
State machine: s0000_Ringing_2_Terminated_X_Reject SIPml-api.js?svn=179:1
=== INVITE Dialog terminated === SIPml-api.js?svn=179:1
PeerConnection::stop() SIPml-api.js?svn=179:1
==stack event = m_permission_accepted SIPml-api.js?svn=179:1
State machine: tsip_transac_ist_Proceeding_2_Completed_X_300_to_699 SIPml-api.js?svn=179:1
SEND: SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 10.168.100.160:5060;rport=5060;branch=z9hG4bK219f1ff4
From: "Josh"<sip:2001@10.168.100.160>;tag=as6beea29a
To: <sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=7lSNF1ZRftqROULGsrJT
Call-ID: 0da24b9c0306805c51ffe7461797a3ed@10.168.100.160:5060
CSeq: 102 INVITE
Content-Length: 0
Reason: SIP; cause=603; text="Failed to get local SDP"

SIPml-api.js?svn=179:1
This/PeerConnection is null: unexpected SIPml-api.js?svn=179:1
==session event = terminated SIPml-api.js?svn=179:1
State machine: tsip_transac_ist_Any_2_Terminated_X_cancel SIPml-api.js?svn=179:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=179:1
recv=ACK sip:tim@df7jal23ls0d.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.168.100.160:5060;rport;branch=z9hG4bK219f1ff4
From: "Josh"<sip:2001@10.168.100.160>;tag=as6beea29a
To: <sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=7lSNF1ZRftqROULGsrJT
Contact: <sip:2001@10.168.100.160:5060;transport=WS>
Call-ID: 0da24b9c0306805c51ffe7461797a3ed@10.168.100.160:5060
CSeq: 102 ACK
Content-Length: 0
Max-Forwards: 70
User-Agent: Asterisk PBX 11.7.0

SIPml-api.js?svn=179:1
SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/WS 10.168.100.160:5060;rport=5060;branch=z9hG4bK219f1ff4
From: "Josh"<sip:2001@10.168.100.160>;tag=as6beea29a
To: <sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=7lSNF1ZRftqROULGsrJT
Call-ID: 0da24b9c0306805c51ffe7461797a3ed@10.168.100.160:5060
CSeq: 102 ACK
Content-Length: 0

SIPml-api.js?svn=179:1
State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister SIPml-api.js?svn=179:1
SEND: REGISTER sip:10.168.100.160 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK7gHgiOL22PZgXZMg6pWH3TqoQASxyWSL;rport
From: "tim"<sip:tim@10.168.100.160>;tag=VTtPO5Zp9wqNFKGflo1B
To: "tim"<sip:tim@10.168.100.160>
Contact: "tim"<sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: c3680078-6129-872f-b2d5-fbe89a7152df
CSeq: 64632 REGISTER
Content-Length: 0
Route: <sip:10.168.100.160:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="tim",realm="asterisk",nonce="3154711a",uri="sip:10.168.100.160",response="ac2eb656ff9ae03942b97ba53ec10be8",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.03.10
Organization: Doubango Telecom

SIPml-api.js?svn=179:1
==session event = sent_request SIPml-api.js?svn=179:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=179:1
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=10.168.107.7;branch=z9hG4bK7gHgiOL22PZgXZMg6pWH3TqoQASxyWSL
From: "tim"<sip:tim@10.168.100.160>;tag=VTtPO5Zp9wqNFKGflo1B
To: "tim"<sip:tim@10.168.100.160>;tag=as1dd684d8
Call-ID: c3680078-6129-872f-b2d5-fbe89a7152df
CSeq: 64632 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.7.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest realm="asterisk",nonce="6a01159c",stale=FALSE,algorithm=MD5

inno2k
Newsterisk
 
Posts: 10
Joined: Tue Mar 25, 2014 4:05 pm

Re: Incoming call to Chrome fails : 603 Failed to get local SDP

Postby navaismo » Tue Mar 25, 2014 9:49 pm

WITHOUT CLI DEBUG!!! only sip debug enabled your debug have a lot of garbage.
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Re: Incoming call to Chrome fails : 603 Failed to get local SDP

Postby inno2k » Wed Mar 26, 2014 10:33 am

Ok I edited the Log post above with CLI Debug off. Thanks for any insights
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Re: Incoming call to Chrome fails : 603 Failed to get local SDP

Postby navaismo » Wed Mar 26, 2014 3:41 pm

SetRemoteDescription failed: Called with a SDP without crypto enabled. SIPml-api.js?svn=179:1


That came from the JS log, how is your webrtc config? Did you enable the secure=yes, icesupport=yes and avpf=yes?

Did you compiled the srtp library before install asterisk?
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*RESOLVED* call to Chrome fails 603 Failed to get local SDP

Postby inno2k » Wed Mar 26, 2014 5:12 pm

Thanks - your comment helped resolve the issue.

I had encryption=yes in sip.conf but I had to add it to each extension as well.

I am now able to receive a call in the browser. Thanks for your help
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Re: *RESOLVED*Call to Chrome fails:603 Failed to get local SDP

Postby mixailom » Sat Jul 12, 2014 3:19 pm

I was following tutorial at http://highsecurity.blogspot.com/2012/1 ... ipml5.html to setup webrtcp call in browser. I used Zoiper to call from one extension another using http://sipml5.org/call.htm?svn=224# in Chrome.
I have all the settings enabled:
transport=udp,ws
encryption=yes
avpf=yes
icesupport=yes

Chrome shows incoming call, when click on allow microphone acces, I still get error in asterisk console: Got SIP response 603 "Failed to get local SDP" back from <IP Address> and call gets rejected.

Using Asterisk 11.5 with srtp enabled.
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Re: *RESOLVED*Call to Chrome fails:603 Failed to get local SDP

Postby navaismo » Sun Jul 13, 2014 5:44 pm

If you're using chrome 35 or superior your error must be the DTLS-SRTP well know issue. Stop posting in every thread, use the forum, google and the jira page to solve it. This is a recurrent issue and has been solved with asterisk patchs and sipml5 work around.
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Re: *RESOLVED*Call to Chrome fails:603 Failed to get local SDP

Postby mixailom » Tue Jul 15, 2014 5:59 am

navaismo wrote:If you're using chrome 35 or superior your error must be the DTLS-SRTP well know issue. Stop posting in every thread, use the forum, google and the jira page to solve it. This is a recurrent issue and has been solved with asterisk patchs and sipml5 work around.


@Navaismo, thank you for your reply. I spent more than a week googling in search for the solution, but I didn't manage to get any page that clearly points way to make webrtc working with asterisk. You were mentioning asterisk patches, but all the patches that are mentioned on forums and some older tutorials are addressing issues from older asterisk versions and older browser versions. I tried both from Chrome 35.0.1916.153 m and Firefox 30 but no success. I (and probably lot of other people with the same issue) would really appreciate if you could post link that has proven working solution. If that would be to bothering to you, if you point me to right direction, I would invest my time and write complete step-by-step tutorial for people in order to get webrtc and asterisk to install from scratch and have it working. I think that open-source community at the moment don't have one that works with current Asterisk and browser versions. Maybe I'm wrong, and I would be happy if there is a link with coplete, working instructions. Thank you in advance.
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Re: *RESOLVED*Call to Chrome fails:603 Failed to get local SDP

Postby navaismo » Tue Jul 15, 2014 7:16 am

@Navaismo, thank you for your reply. I spent more than a week googling in search for the solution, but I didn't manage to get any page that clearly points way to make webrtc working with asterisk.

I have worked for 4 months before my first call between Chrome and my asterisk, and then 4 months to integrate the webrtc to FreePBX and Elastix.

You were mentioning asterisk patches, but all the patches that are mentioned on forums and some older tutorials are addressing issues from older asterisk versions and older browser versions. I tried both from Chrome 35.0.1916.153 m and Firefox 30 but no success. I (and probably lot of other people with the same issue) would really appreciate if you could post link that has proven working solution.

Already did that: viewtopic.php?f=1&t=90167

Your issue seems to fit in the last Issue added on June 4 of that link.

A huge work was done in the Jira page https://issues.asterisk.org/jira/browse/ASTERISK-22961 to resolve another ones.


If that would be to bothering to you, if you point me to right direction, I would invest my time and write complete step-by-step tutorial for people in order to get webrtc and asterisk to install from scratch and have it working. I think that open-source community at the moment don't have one that works with current Asterisk and browser versions. Maybe I'm wrong, and I would be happy if there is a link with coplete, working instructions. Thank you in advance.

Me and others already spent a lot of time helping the community with webrtc you can check the asterisk IRC log, my blog or my github repo. But people seems to be lazy to use google and find the solution by yourself. I´m not complaining but if you see the the IRC logs ar how many threads about asterisk+webrtc are opened daily with the same issues you will found that exhausting.

Developers from JsSIP doesn´t provide Help about their API with asterisk because repetitive threads, folks on doubango already provided a temporary patch and instruccions to use the media gateway with DTLS support.

Asterisk Developers are already tired of this too --> http://www.joshua-colp.com/webrtc-let-m ... t-on-that/

So please, before flooding the IRC or the Forum do a deeper search, a week spent is like nothing since most of us are trying to work with this since the past year do the math and calculate our free time spent for the community.

If you want to work like the most of users without go deeper let me tell you something: Go download Elastix 2.4 install my WebRTC addon(WebRTC Agent Console) then use google translate to understand my tutorial here http://asterisktools.blogspot.com/2013/ ... e-con.html

And finally apply this patch http://forum.elastix.org/viewtopic.php?f=18&t=128877 to the Elastix system and use it until Chrome 37 finally break everything.

Best Regards.
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Re: *RESOLVED*Call to Chrome fails:603 Failed to get local SDP

Postby kair_bek » Thu Jul 17, 2014 2:16 am

Hi navaismo,

I appreciate your contribution to the community. Most of your entries were helpful for me
while troubleshooting the problems, especially related to WebRTC.
Currently, I am trying to make a call browser-to-softphone via Asterisk. And actually it is done on Ubuntu 14.04 with Asterisk 11.11.0. Now I need to do the same on other machine, which is Elastix 2.4 distro. I would like to know, is it ok upgrading the Asterisk on Elastix by building Asterisk from the source? Or there is another more preferred way? Would you mind if I ask you to see my real problem here viewtopic.php?f=1&t=90890.
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