Failed to get Local SDP

Jay

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Jan 27, 2014, 7:14:15 PM1/27/14

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Initially started as a topic here, and referred to Doubango by Asterisk team.

Asterisk 11.7 + SIPML5 + WSS, both Firefox and Chrome (not using WebRTC2SIP).

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.

Reliably Transmitting (NAT) to 1.1.1.1.1:22596:
INVITE sip:10...@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:10...@2.2.2.2>;tag=as07bcc9b2
To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:10...@2.2.2.2:5060;transport=WS>
Call-ID: 654bc8580334ada5...@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

<--- SIP read from WS:1.1.1.1:22596 --->

SIP/2.0 180 Ringing
Via: SIP/2.0/WS 2.2.2.2:5060;rport=5060;branch=z9hG4bK5d6845a4
From: "New User"<sip:10...@2.2.2.2>;tag=as07bcc9b2
To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA
Contact: <sip:10...@df7jal23ls0d.invalid;transport=ws>
Call-ID: 654bc8580334ada5...@2.2.2.2:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
<------------->
— (9 headers 0 lines) —
list_route: hop: <sip:10...@df7jal23ls0d.invalid;transport=ws>
– SIP/1061-00000015 is ringing
<--- SIP read from WS:1.1.1.1:22596 --->
SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 2.2.2.2:5060;rport=5060;branch=z9hG4bK5d6845a4
From: "New User"<sip:10...@2.2.2.2>;tag=as07bcc9b2
To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA
Call-ID: 654bc8580334ada5...@2.2.2.2:5060
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 1.1.1.1:22596
set_destination: Parsing <sip:10...@df7jal23ls0d.invalid;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Transmitting (NAT) to 1.1.1.1:22596:
ACK sip:10...@df7jal23ls0d.invalid;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:10...@2.2.2.2>;tag=as07bcc9b2
To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA
Contact: <sip:10...@2.2.2.2:5060;transport=WS>
Call-ID: 654bc8580334ada5...@12.2.2.2:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0-rc1
Content-Length: 0

We are aware that this issue may be solved using WebRTC2SIP. However, we have decided to take on the journey of a proxy-less set-up, so any help would be appreciated.

navaismo

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Jan 31, 2014, 10:17:06 AM1/31/14

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Show us the SIP settings

El lunes, 27 de enero de 2014 18:14:15 UTC-6, Jay escribió:

Initially started as a topic here, and referred to Doubango by Asterisk team.

Asterisk 11.7 + SIPML5 + WSS, both Firefox and Chrome (not using WebRTC2SIP).

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.

Reliably Transmitting (NAT) to 1.1.1.1.1:22596:
INVITE sip:10...@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:...@2.2.2.2>;tag=as07bcc9b2

From: "New User"<sip:...@2.2.2.2>;tag=as07bcc9b2

To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA
Contact: <sip:10...@df7jal23ls0d.invalid;transport=ws>
Call-ID: 654bc8580334ada5...@2.2.2.2:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
<------------->
— (9 headers 0 lines) —
list_route: hop: <sip:10...@df7jal23ls0d.invalid;transport=ws>
– SIP/1061-00000015 is ringing
<--- SIP read from WS:1.1.1.1:22596 --->
SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 2.2.2.2:5060;rport=5060;branch=z9hG4bK5d6845a4

From: "New User"<sip:...@2.2.2.2>;tag=as07bcc9b2

To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA
Call-ID: 654bc8580334ada5...@2.2.2.2:5060
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 1.1.1.1:22596
set_destination: Parsing <sip:10...@df7jal23ls0d.invalid;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Transmitting (NAT) to 1.1.1.1:22596:
ACK sip:10...@df7jal23ls0d.invalid;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:...@2.2.2.2>;tag=as07bcc9b2

To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA
Contact: <sip:10...@2.2.2.2:5060;transport=WS>
Call-ID: 654bc8580334ada5...@12.2.2.2:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0-rc1
Content-Length: 0

shyam sundar kulkarni

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Feb 2, 2014, 3:58:44 PM2/2/14

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It means the payload type mismatch, Please see what's there in your config.xml. Change this attribute to YES.

<enable-media-coder>yes</enable-media-coder>

you should not see this error further.

On Tuesday, 28 January 2014 05:44:15 UTC+5:30, Jay wrote:

Initially started as a topic here, and referred to Doubango by Asterisk team.

Asterisk 11.7 + SIPML5 + WSS, both Firefox and Chrome (not using WebRTC2SIP).

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.

Reliably Transmitting (NAT) to 1.1.1.1.1:22596:
INVITE sip:10...@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:...@2.2.2.2>;tag=as07bcc9b2

To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:10...@2.2.2.2:5060;transport=WS>

Call-ID: 654bc8580334ada51c6cd2d75b83f46...@162.209.101.19:5060

From: "New User"<sip:...@2.2.2.2>;tag=as07bcc9b2

To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA
Contact: <sip:10...@df7jal23ls0d.invalid;transport=ws>

Call-ID: 654bc8580334ada51c6cd2d75b83f46...@2.2.2.2:5060

CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
<------------->
— (9 headers 0 lines) —
list_route: hop: <sip:10...@df7jal23ls0d.invalid;transport=ws>
– SIP/1061-00000015 is ringing
<--- SIP read from WS:1.1.1.1:22596 --->
SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 2.2.2.2:5060;rport=5060;branch=z9hG4bK5d6845a4

From: "New User"<sip:...@2.2.2.2>;tag=as07bcc9b2

To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA

Call-ID: 654bc8580334ada51c6cd2d75b83f46...@2.2.2.2:5060

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 1.1.1.1:22596
set_destination: Parsing <sip:10...@df7jal23ls0d.invalid;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Transmitting (NAT) to 1.1.1.1:22596:
ACK sip:10...@df7jal23ls0d.invalid;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:...@2.2.2.2>;tag=as07bcc9b2

To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA
Contact: <sip:10...@2.2.2.2:5060;transport=WS>

Call-ID: 654bc8580334ada51c6cd2d75b83f46...@12.2.2.2:5060

CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0-rc1
Content-Length: 0

Jay

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Feb 3, 2014, 3:19:03 PM2/3/14

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We are not using WEBRTC2SIP, and thus have no such option...

Jay

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Feb 3, 2014, 3:19:33 PM2/3/14

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Additional info that my help in troubleshooting this issue.

We are trying to establish a DTLS-SRTP-encrypted connection. On the outgoing call, we can see the following in the INVITE:

m=audio 50152 UDP/TLS/RTP/SAVPF 109 0 8 101

However, when Asterisk is bridging the connection, it sends the following INVITE:

m=audio 19798 RTP/SAVPF 0 3 8 101

However, SIPML5 is expecting UDP/TLS/RTP/SAVPF in the "m=" field.

Therefore, SIPML5 refuses the cooection with 603 Failed to get local SDP.

More info here: http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/61967

On Monday, January 27, 2014 7:14:15 PM UTC-5, Jay wrote:

Initially started as a topic here, and referred to Doubango by Asterisk team.

Asterisk 11.7 + SIPML5 + WSS, both Firefox and Chrome (not using WebRTC2SIP).

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.

Reliably Transmitting (NAT) to 1.1.1.1.1:22596:
INVITE sip:10...@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:10...@2.2.2.2>;tag=as07bcc9b2
To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:10...@2.2.2.2:5060;transport=WS>

Call-ID: 654bc8580334ada51c6cd2d75b83f46...@162.209.101.19:5060

Call-ID: 654bc8580334ada51c6cd2d75b83f46...@2.2.2.2:5060

CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
<------------->
— (9 headers 0 lines) —
list_route: hop: <sip:10...@df7jal23ls0d.invalid;transport=ws>
– SIP/1061-00000015 is ringing
<--- SIP read from WS:1.1.1.1:22596 --->
SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 2.2.2.2:5060;rport=5060;branch=z9hG4bK5d6845a4
From: "New User"<sip:10...@2.2.2.2>;tag=as07bcc9b2
To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA

Call-ID: 654bc8580334ada51c6cd2d75b83f46...@2.2.2.2:5060

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 1.1.1.1:22596
set_destination: Parsing <sip:10...@df7jal23ls0d.invalid;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Transmitting (NAT) to 1.1.1.1:22596:
ACK sip:10...@df7jal23ls0d.invalid;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:10...@2.2.2.2>;tag=as07bcc9b2
To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA
Contact: <sip:10...@2.2.2.2:5060;transport=WS>

Call-ID: 654bc8580334ada51c6cd2d75b83f46...@12.2.2.2:5060

CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0-rc1
Content-Length: 0

Jay

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Feb 3, 2014, 3:29:47 PM2/3/14

to doub...@googlegroups.com

I have attached the debug dump. Hope this helps.

On Monday, January 27, 2014 7:14:15 PM UTC-5, Jay wrote:

Initially started as a topic here, and referred to Doubango by Asterisk team.

Asterisk 11.7 + SIPML5 + WSS, both Firefox and Chrome (not using WebRTC2SIP).

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.

Reliably Transmitting (NAT) to 1.1.1.1.1:22596:
INVITE sip:10...@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:10...@2.2.2.2>;tag=as07bcc9b2
To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:10...@2.2.2.2:5060;transport=WS>

Call-ID: 654bc8580334ada51c6cd2d75b83f46...@162.209.101.19:5060

Call-ID: 654bc8580334ada51c6cd2d75b83f46...@2.2.2.2:5060

CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
<------------->
— (9 headers 0 lines) —
list_route: hop: <sip:10...@df7jal23ls0d.invalid;transport=ws>
– SIP/1061-00000015 is ringing
<--- SIP read from WS:1.1.1.1:22596 --->
SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 2.2.2.2:5060;rport=5060;branch=z9hG4bK5d6845a4
From: "New User"<sip:10...@2.2.2.2>;tag=as07bcc9b2
To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA

Call-ID: 654bc8580334ada51c6cd2d75b83f46...@2.2.2.2:5060

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 1.1.1.1:22596
set_destination: Parsing <sip:10...@df7jal23ls0d.invalid;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Transmitting (NAT) to 1.1.1.1:22596:
ACK sip:10...@df7jal23ls0d.invalid;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:10...@2.2.2.2>;tag=as07bcc9b2
To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA
Contact: <sip:10...@2.2.2.2:5060;transport=WS>

Call-ID: 654bc8580334ada51c6cd2d75b83f46...@12.2.2.2:5060

CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0-rc1
Content-Length: 0

Dump.txt.txt

Mamadou DIOP

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Feb 3, 2014, 4:23:00 PM2/3/14

to doub...@googlegroups.com

DTLS must be used with UDP/TLS/RTP/SAVPF (rfc5764) which means what SIPML is doing is correct. I'd say that chrome, firefox and asterisk are all buggy unless I've missed something.

SIPML can accept both UDP/TLS/RTP/SAVPF and RTP/SAVPF which means your diagnostic is not correct.

You should provide clear technical description and clean trace for both Asterisk and the browser if you want help.

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<Dump.txt.txt>

Jay

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Feb 4, 2014, 10:40:04 PM2/4/14

to doub...@googlegroups.com

Mamadou,

Although the issue may be on the Asterisk side, we are trying a parralel approach, seeking the issue on both ends. Hopefully that will speed up the process.

Some believe that it may be caused by SIPML (or can be solved by modifying SIPML easier), that being the reason for me to reach out.

Here is the relevant Digium JIRA Issue: https://issues.asterisk.org/jira/browse/ASTERISK-23191

I have attached the sip debug log.

In addition, here are my rtp and sip configs:

rtp.conf

[general]

rtpstart=10000
rtpend=20000

icesupport=true
stunaddr=stun.l.google.com:19302

users.conf

[1060]
type=peer
defaultuser=1060
host = dynamic
secret=password
context=default
autoprov = yes
allow=all
hasiax=no
hassip = yes
nat=force_rport
avpf = yes
encryption = yes
icesupport=yes
videosupport=no
directmedia=no
dtlsenable = yes
dtlsverify = no
dtlscertfile=/etc/ssl/cert/device/device.crt
dtlsprivatekey=/etc/ssl/cert/device/device.key
dtlscafile=/etc/ssl/cert/rootCA.pem

[1061]
type=peer
defaultuser=1061
host=dynamic
secret=anotherpassword
context=default
hasiax = no
hassip = yes
allow=all
nat=force_rport
encryption = yes
avpf = yes
icesupport = yes
videosupport=no
directmedia=no
dtlsenable = yes
dtlsverify = no
dtlscertfile=/etc/ssl/cert/device/device.crt
dtlsprivatekey=/etc/ssl/cert/device/device.key
dtlscafile=/etc/ssl/cert/rootCA.pem

On Monday, January 27, 2014 7:14:15 PM UTC-5, Jay wrote:

Initially started as a topic here, and referred to Doubango by Asterisk team.

Asterisk 11.7 + SIPML5 + WSS, both Firefox and Chrome (not using WebRTC2SIP).

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.

Reliably Transmitting (NAT) to 1.1.1.1.1:22596:
INVITE sip:10...@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:10...@2.2.2.2>;tag=as07bcc9b2
To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:10...@2.2.2.2:5060;transport=WS>

Call-ID: 654bc8580334ada51c6cd2d75b83f46...@162.209.101.19:5060

Call-ID: 654bc8580334ada51c6cd2d75b83f46...@2.2.2.2:5060

CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
<------------->
— (9 headers 0 lines) —
list_route: hop: <sip:10...@df7jal23ls0d.invalid;transport=ws>
– SIP/1061-00000015 is ringing
<--- SIP read from WS:1.1.1.1:22596 --->
SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 2.2.2.2:5060;rport=5060;branch=z9hG4bK5d6845a4
From: "New User"<sip:10...@2.2.2.2>;tag=as07bcc9b2
To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA

Call-ID: 654bc8580334ada51c6cd2d75b83f46...@2.2.2.2:5060

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 1.1.1.1:22596
set_destination: Parsing <sip:10...@df7jal23ls0d.invalid;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Transmitting (NAT) to 1.1.1.1:22596:
ACK sip:10...@df7jal23ls0d.invalid;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:10...@2.2.2.2>;tag=as07bcc9b2
To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA
Contact: <sip:10...@2.2.2.2:5060;transport=WS>

Call-ID: 654bc8580334ada51c6cd2d75b83f46...@12.2.2.2:5060

CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0-rc1
Content-Length: 0

Jay

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Feb 4, 2014, 10:44:33 PM2/4/14

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It won't let me attach a file for some strange reason, but here it is: https://www.dropbox.com/s/ftk8qdy4hga6f5p/debuglog-cleared-ip.txt

Mamadou DIOP

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Feb 4, 2014, 10:59:00 PM2/4/14

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Some believe that it may be caused by SIPML (or can be solved by modifying SIPML easier), that being the reason for me to reach out.

Doesn't make sense. Who are "some"? Why you think issue is in SIPML?

I already said there is an issue in Asterisk (invalid profile), the new trace (see bellow) you posted shows that there is another issue: The SDP uses "RTP/SAVPF" profile but you don't have neither "a:fingerprint" nor "a:crypto".

Instead of trying to guess what's the issue, you should forward what I've said to digium guys.

ÿINVITE sips:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss SIP/2.0
ÿVia: SIP/2.0/WS 1.1.1.1:5060;branch=z9hG4bK212b468c;rport
ÿMax-Forwards: 70
ÿFrom: "New User" <sip:10...@1.1.1.1>;tag=as5c54e108
ÿTo: <sips:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>
ÿContact: <sip:10...@1.1.1.1:5060;transport=WS>
ÿCall-ID: 477df3784220ff3e...@1.1.1.1:5060
ÿCSeq: 102 INVITE
ÿUser-Agent: Asterisk PBX 11.7.0
ÿDate: Tue, 04 Feb 2014 05:40:24 GMT
ÿAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
ÿSupported: replaces, timer
ÿContent-Type: application/sdp
ÿContent-Length: 2077
ÿ
ÿv=0
ÿo=root 691573191 691573191 IN IP4 1.1.1.1
ÿs=Asterisk PBX 11.7.0
ÿc=IN IP4 1.1.1.1
ÿt=0 0
ÿm=audio 16156 RTP/SAVPF 0 4 3 3 8 8 112 5 7 18 110 97 111 9 102 115 116 117 96 100 107 108 10 118 119 101
ÿa=rtpmap:0 PCMU/8000
ÿa=rtpmap:4 G723/8000
ÿa=fmtp:4 annexa=no
ÿa=rtpmap:3 GSM/8000
ÿa=rtpmap:3 GSM/8000
ÿa=rtpmap:8 PCMA/8000
ÿa=rtpmap:8 PCMA/8000
ÿa=rtpmap:112 AAL2-G726-32/8000
ÿa=rtpmap:5 DVI4/8000
ÿa=rtpmap:7 LPC/8000
ÿa=rtpmap:18 G729/8000
ÿa=fmtp:18 annexb=no
ÿa=rtpmap:110 speex/8000
ÿa=rtpmap:97 iLBC/8000
ÿa=fmtp:97 mode=30
ÿa=rtpmap:111 G726-32/8000
ÿa=rtpmap:9 G722/8000
ÿa=rtpmap:102 G7221/16000
ÿa=fmtp:102 bitrate=32000
ÿa=rtpmap:115 G7221/32000
ÿa=fmtp:115 bitrate=48000
ÿa=rtpmap:116 G719/48000
ÿa=fmtp:116 bitrate=64000
ÿa=rtpmap:117 speex/16000
ÿa=rtpmap:96 SILK/8000
ÿa=fmtp:96 maxaveragebitrate=10000
ÿa=fmtp:96 usedtx=0
ÿa=fmtp:96 useinbandfec=1
ÿa=rtpmap:100 SILK/12000
ÿa=fmtp:100 maxaveragebitrate=12000
ÿa=fmtp:100 usedtx=0
ÿa=fmtp:100 useinbandfec=1
ÿa=rtpmap:107 SILK/16000
ÿa=fmtp:107 maxaveragebitrate=20000
ÿa=fmtp:107 usedtx=0
ÿa=fmtp:107 useinbandfec=1
ÿa=rtpmap:108 SILK/24000
ÿa=fmtp:108 maxaveragebitrate=30000
ÿa=fmtp:108 usedtx=0
ÿa=fmtp:108 useinbandfec=1
ÿa=rtpmap:10 L16/8000
ÿa=rtpmap:118 L16/16000
ÿa=rtpmap:119 speex/32000
ÿa=rtpmap:101 telephone-event/8000
ÿa=fmtp:101 0-16
ÿa=silenceSupp:off - - - -
ÿa=ptime:20
ÿa=ice-ufrag:31b1f27c2426a56a68fdc51b516682f6
ÿa=ice-pwd:4c33d4467e084b9c1682dce1040c5d80
ÿa=candidate:Ha2f2df83 1 UDP 2130706431 1.1.1.1 16156 typ host
ÿa=candidate:Hab0a297 1 UDP 2130706431 10.176.162.151 16156 typ host
ÿa=candidate:Hc0a8a804 1 UDP 2130706431 192.168.168.4 16156 typ host
ÿa=candidate:Sa2f2df83 1 UDP 1694498815 1.1.1.1 16156 typ srflx
ÿa=candidate:Ha2f2df83 2 UDP 2130706430 1.1.1.1 16157 typ host
ÿa=candidate:Hab0a297 2 UDP 2130706430 10.176.162.151 16157 typ host
ÿa=candidate:Hc0a8a804 2 UDP 2130706430 192.168.168.4 16157 typ host
ÿa=candidate:Sa2f2df83 2 UDP 1694498814 1.1.1.1 16157 typ srflx
ÿa=connection:new
ÿa=setup:active
ÿa=sendrecv
ÿ

Jay

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Feb 5, 2014, 8:43:33 PM2/5/14

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"Failed to get Local SDP" is coming from SIPml5, that's why we're trying to figure out what might be the solution for this behavior on SIPML side.

Firefox shows both fingerprint and crypto, which is absent for chrome, for some reason (during a call from Chrome to Firefox).

I have created an additional log for both firefox (callee) and chrome (caller) at the same time. Here it is: https://www.dropbox.com/sh/hg0l1t153kz1fcq/zpeKCDkp7y

This illustrates the strange issue with accepting the call on the receiving side (while the caller is being connected).

On Tuesday, February 4, 2014 10:59:00 PM UTC-5, Mamadou wrote:

Some believe that it may be caused by SIPML (or can be solved by modifying SIPML easier), that being the reason for me to reach out.
Doesn't make sense. Who are "some"? Why you think issue is in SIPML?
I already said there is an issue in Asterisk (invalid profile), the new trace (see bellow) you posted shows that there is another issue: The SDP uses "RTP/SAVPF" profile but you don't have neither "a:fingerprint" nor "a:crypto".

Instead of trying to guess what's the issue, you should forward what I've said to digium guys.

ÿINVITE sips:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss SIP/2.0
ÿVia: SIP/2.0/WS 1.1.1.1:5060;branch=z9hG4bK212b468c;rport
ÿMax-Forwards: 70
ÿFrom: "New User" <sip:10...@1.1.1.1>;tag=

as5c54e108
ÿTo: <sips:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>
ÿContact: <sip:10...@1.1.1.1:5060;transport=WS>
ÿCall-ID: 477df3784220ff3e1e96db5a4958aab...@1.1.1.1:5060

From: "New User" <sip:...@2.2.2.2>;tag=as07bcc9b2

From: "New User"<sip:...@2.2.2.2>;tag=as07bcc9b2

To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA
Contact: <sip:10...@df7jal23ls0d.invalid;transport=ws>
Call-ID: 654bc8580334ada51c6cd2d75b83f46...@2.2.2.2:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
<------------->
— (9 headers 0 lines) —
list_route: hop: <sip:10...@df7jal23ls0d.invalid;transport=ws>
– SIP/1061-00000015 is ringing
<--- SIP read from WS:1.1.1.1:22596 --->
SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 2.2.2.2:5060;rport=5060;branch=z9hG4bK5d6845a4

From: "New User"<sip:...@2.2.2.2>;tag=as07bcc9b2

To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA
Call-ID: 654bc8580334ada51c6cd2d75b83f46...@2.2.2.2:5060
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 1.1.1.1:22596
set_destination: Parsing <sip:10...@df7jal23ls0d.invalid;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Transmitting (NAT) to 1.1.1.1:22596:
ACK sip:10...@df7jal23ls0d.invalid;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:...@2.2.2.2>;tag=as07bcc9b2

To: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=NfQp0QFtldX32ZlHqLGA
Contact: <sip:10...@2.2.2.2:5060;transport=WS>
Call-ID: 654bc8580334ada51c6cd2d75b83f46...@12.2.2.2:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0-rc1
Content-Length: 0

We are aware that this issue may be solved using WebRTC2SIP. However, we have decided to take on the journey of a proxy-less set-up, so any help would be appreciated.

Mamadou DIOP

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Feb 5, 2014, 8:58:31 PM2/5/14

to doub...@googlegroups.com

I don't really know how I have to explain it clearly.

I'll repeat it again: The SDP from Asterisk IS NOT CORRECT. Nothing to do with SIPML. SIPML is returning "Failed to get local SDP" because the WebRTC doesn't accept the incoming SDP and there is a good reason as it's incorrect.

I don't know if it's a bug in Asterisk or in your config.

Jay

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Feb 6, 2014, 2:56:25 PM2/6/14

to doub...@googlegroups.com

Thanks, we'll keep digging then.

On a separate note - were you ever able to make SIPML WSS calls without webrtc2sip? If so, that would mean that it's our configuration that's causing the issue. If not, then it's most likely that this is an Asterisk issue, and thus a patch on that side is necessary.

Thanks.

Jay

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Feb 27, 2014, 7:24:52 PM2/27/14

to doub...@googlegroups.com

Further troubleshooting has indicated that this may in fact be a SIPML5 issue.

Please see this topic: https://issues.asterisk.org/jira/browse/ASTERISK-22961

A JS hack to replace RTP/SAVPF in SDP with UDP/TLS/RTP/SAVPF seems to be working.

"Asterisk is correct in its behaviour. For DTLS-SRTP, media profile should be UDP/TLS/RTP/SAVP(F). "

Mamadou DIOP

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Feb 27, 2014, 7:46:18 PM2/27/14

to doub...@googlegroups.com

This has been discussed several times and I reported it to both Mozilla and Google a year ago.

Chrome: we use SDES which means RTP/SAVPF is correct

FF: we use DTLS which means UDP/TLS/RTP/SAVPF (patched in JS code) is correct.

In short, there is no bug in SIPML5. We have many commercial products using SIPML5 (both chrome and FF) to call Asterisk and all is working well.

Jay

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Mar 5, 2014, 2:02:50 PM3/5/14

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This has been changed for Chrome, which now defaults to DTLS, and SDES will be removed fully in the upcoming version of Chrome https://code.google.com/p/webrtc/issues/detail?id=2774

Thus I believe it is time to update that in SIPML as well, as SDES for WebRTC will be discontinued very soon (as per IETF's decision).

More on that: https://groups.google.com/forum/#!topic/discuss-webrtc/67W4zrTmoNs

Jay

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Mar 17, 2014, 12:23:18 PM3/17/14

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Hi Mamadou,

Could you please let me know if this change is planned to be released in the nearest future?

Alternatively, could you guide me to the right place to adjust that behavior? I assume the adjustments have to be done in tmedia_session_jsep.js? If so, where can I learn more about the variables used there as they are quite confusing for us to work with?

Thanks a lot.

Brian Capouch

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Jul 6, 2014, 2:01:37 PM7/6/14

to doub...@googlegroups.com

Jay have you been able to get this going after the SHA-256 patch was added to Asterisk? I have applied the patch and incoming calls to siipML still fail in the precise way indicated here. Asterisk issue 23191 pertains to this problem, and was marked "Closed, Duplicate" to the issue solved with the SHA-256 patch. Obviously, the two issues are separate.

It looks like both sides are blaming the other as the cause of the problem. If there is a workaround you know of, I'd sure be obliged to learn it.

Once upon a long time ago, in the days of SDES, things worked OK for incoming calls.

Mikkel Bach-Mortensen

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Jul 7, 2014, 11:01:54 AM7/7/14

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Seriously, switch to web breaker (webrtc2sip), it's stable and just works. Let google, firefox and Asterisk battle their blame game and enjoy a working system instead.

Gopalakrishnan N

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Jul 9, 2014, 5:12:30 AM7/9/14

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Hi,

Even after enabling the RTC Web Tracker am getting failed to get SDP.

The first invite i have like this "INVITE sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0"

I tried dialing from eyebeam to sipml5.

When i try to dial from sipml5 to eyebeam same dialog but different error message in Asterisk CLI

"chan_sip.c:10454 process_sdp: Rejecting secure audio stream without encryption details: audio 46695 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126"

Any one has any fix for this?

Regards

Mikkel Bach-Mortensen

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Jul 9, 2014, 1:25:10 PM7/9/14

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Well you haven't installed a webrtc2sip (or done so correctly) your asterisk should be talking plain unencrypted rtp between it and the webrtc2sip.

You received this message because you are subscribed to a topic in the Google Groups "discuss-doubango" group.
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--

Venlig hilsen / Best regards
Mikkel Bach-Mortensen

CTO, Bellmetric ApS
Direkte tlf. 7879 2225
E-mail: mik...@bellmetric.net

http://www.bellmetric.net

Message has been deleted

Gopalakrishnan N

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Jul 21, 2014, 9:32:05 AM7/21/14

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Sorry for my delayed reply, let me try to install this. Thank you.

Regards

Mikkel Bach-Mortensen

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Jul 21, 2014, 9:42:24 AM7/21/14

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Yes, follow that guide for installing webrtc2sip (and doubango).

Now an important thing to understand, the setup becomes:

Client <-dtls-> webrtc2sip <-sip-> asterisk

You need proper certificates for the webrtc2sip. The asterisk in this setup should *not* be talking encrypted, ice or anything like that.

Your WSS address is the webrtc2sip, your sip registration should be to the asterisk (and needs to only be accessible from the webrtc2sip, not the client). Remember to enable web breaker in the client.

Gopalakrishnan N

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Jul 21, 2014, 10:42:00 AM7/21/14

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webrtc2sip and Asterisk can be in both same machine right?

And to create certificate am going to follow this thread - https://groups.google.com/forum/#!topic/doubango/asAfP5ZCgdI

Thanks.

Mikkel Bach-Mortensen

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Jul 22, 2014, 2:47:23 AM7/22/14

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In theory yes, however, you probably should modify the source code of webrtc2sip to only allocate a certain set of ports, and have asterisk work on a different set of ports.

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Gopalakrishnan N

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Jul 28, 2014, 4:48:14 AM7/28/14

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Hi,

Finally I installed webrtc2sip with other required modules. And have created certificates "self signed certificate" configured the same in config.xml.

I have changed the SIP port to 5070 in my Asterisk, where webrtc2sip also installed in the same machine.

Post to these, when I do a outbound call from browser (Chrome/Firefox), I still get the error "MSG: Remote party requesting DTLS-DTLS (UDP/TLS/RTP/SAVPF) but this option is not enabled" and while incoming from a softphone to browser, I get the "Got SIP response 603 "Failed to get local SDP" back from 10.20.254.131:10060"

Basically the RTP is not communicating between the clients.

I see in my Asterisk CLI prompt where the browser client is getting registered with the same IP where Asterisk is installed, my thought here is; the RTP is looping back to the same server itself? or Shall I try to have Asterisk in different machine?

Thanks.

Gopalakrishnan N

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Jul 28, 2014, 6:47:18 AM7/28/14

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I got little improvement, where in webrtc2sip console there are no errors, and I think I have to configure Asterisk side, where I am getting to setup SRTP policies in Asterisk CLI prompt.

"sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP policies"

Regards

Mikkel Bach-Mortensen

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Jul 28, 2014, 7:58:18 AM7/28/14

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Still sounds like you are trying to talk encrypted from asterisk.

You need to post your sip.conf (remember to remove passwords!) and rtp.conf from asterisk and your webrtc2sip config.

Also need your sip client setup (and the output from call.html in console).

/Mikkel

Gopalakrishnan N

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Jul 28, 2014, 10:09:52 AM7/28/14

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Please find the attached files.

And in the webrtc2sip console, I didn't get any error.

[root@testserver webrtc2sip]# /opt/webrtc2sip/sbin/webrtc2sip

*******************************************************************

PRODUCT: webrtc2sip

HOME PAGE: http://webrtc2sip.org

LICENCE: GPLv3 or proprietary

VERSION: 2.6.0

'quit' to quit the application.

*******************************************************************

SSL is enabled :)

DTLS supported: yes

DTLS-SRTP supported: yes

call_html.xlsx

config.xml

rtp.conf

sip.conf

sip_test.conf

Gopalakrishnan N

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Jul 28, 2014, 11:58:23 AM7/28/14

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Whether a=crypto has to come in SDP, I think because of this, my RTP audio is rejecting,

" WARNING[1867][C-0000000b]: chan_sip.c:10454 process_sdp: Rejecting secure audio stream without encryption details: audio 35669 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126"

Regards

Gopalakrishnan N

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Jul 28, 2014, 12:13:14 PM7/28/14

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Sorry to keep on posting again and again,

Now I have the crypto value in my SDP,

a=acap:5 crypto:1 AES_CM_128_HMAC_SHA1_80 inline:53KfpcLzY/5RU4Ug16UIi0/PQYL6KfCufKxXyryj

a=acap:6 crypto:2 AES_CM_128_HMAC_SHA1_32 inline:0Gim+YMosWRk7h8uuGqLw1r+Jt23T4EHVltIWxrs

but even though I have enabled avpf=yes in both of my extensions in sip_test.conf, I get the following error,

[Jul 28 21:40:23] WARNING[1867][C-00000016]: chan_sip.c:10124 process_sdp: Received AVP profile in audio offer but AVPF is enabled: audio 57384 RTP/AVP 111 8 0 101

[Jul 28 21:40:23] WARNING[1867][C-00000016]: chan_sip.c:10473 process_sdp: Failing due to no acceptable offer found

Gopalakrishnan N

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Jul 28, 2014, 10:54:44 PM7/28/14

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Do I need to use my SIP peer like this,


[6001]
host=dynamic
secret=DONT_USE_THIS_INSECURE_PASSWORD
context=from-internal
type=friend
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass

Whether I need to create DTLS certificate file in Asterisk?

Thanks.

Thank you with regards,
Gopalkrishnan N.

Mob: +91 99404 91346

VoIP call - sip:sai...@gtalk2voip.com

Gopalakrishnan N

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Jul 29, 2014, 5:26:42 AM7/29/14

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Hi Mikkel,

Thanks for your kindly comments, after removing encryption=yes from SIP extension, am able to make it work with chrome both inbound and outbound.

Regards,

Gopal.

Thank you with regards,
Gopalkrishnan N.

Mob: +91 99404 91346

VoIP call - sip:sai...@gtalk2voip.com

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