[asterisk-users] PJSIP Real-time Text (T.140)

[Simon Hohberg]

Hi,

is the support of real-time text limited to the SIP channel driver only? 
Somehow Asterisk is not offering T.140 to the called party when 
initiating a call and including real-time text.


In my pjsip.conf I allowed T.140 and enabled text support.


Regards,

Simon Hohberg

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[asterisk-users] Empty user string on pjsip inbound trunk

[Simon Hohberg]

Hi,

I try to setup an inbound trunk using pjsip_wizard.conf. Now, when I 
receive a call from that trunk with an empty user string, and I try to 
match it with 's' in the dial plan, Asterisk reports that the extension 
was not found in the context.


* pjsip_wizard.conf:

[example]
type = wizard
sends_auth = no
sends_registrations = no
remote_hosts = example.com:5060
endpoint/context = from-extern

* extensions.conf:

[from-extern]
exten => s,1,Playback(demo-thanks)

* Asterisk log:

res_pjsip_session.c: Call from 'example' (UDP:111.222.3.4:5060) to 
extension '' rejected because extension not found in context 'from-extern'.



What am I doing wrong?


Regards,

Simon

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Re: [asterisk-users] PJSIP - Video Support for WebRTC

[Simon Hohberg]

On 07/26/2016 03:15 PM, Olivier wrote:

Matthew Jordan  digium.com> writes:



On Mon, Mar 23, 2015 at 8:55 AM, Gosmac  gmail.com>

wrote:

Hey i have an interesting topic to discuss here.

The main goal here is to be able to make a video call between two

WebRTC endpoints registered on asterisk 13

it is a feature that definitely asterisk 13 should support .


the problems that i faced with this is the following and i hope i

could get an advise here.


asterisk 13 vanilla version has some issues marking the video

packets this complain web browser

specially VP8 codecs so a friend of mine help me to patch

res_rtp_asterisk and now asterisk is marking

video streams :) it just mark video packets not touch anything else

and web browser show video on web page

now I’m using online demo http://tryit.jssip.net/ is stable and get

more updates than sipml5. so i try

echo() dialplan test and everything work perfect on echo test :).


i have two questions and i hope you could give me some advise.

1) after marking video packet I’m able to make Dial() between two

webrtc peers but i get one way audio and

video on callee party, “after 3 minutes on call” i get two way audio

and video on all parties seems to be

not just a problem on a missing keyframe.


 1.1) the 3 minutes delay only happen using chrome stable , could be

a dtls problem when asterisk make an

offer to other endpoint?

 1.2) when i use chrome-dev and i disable dlts encryption everything

work perfect on video call.


2) after marking video packets i realize that when you make a call

with video and you involve on dialplan an

application like playback or music on hold any application that

played audio files (audio and video never work).


2.1) asterisk is muggling the audio and video streams ?

This is good information for all guys out there that wants to

support video on webrtc in asterisk 13




Please stop spamming the list with this e-mail. Resending it multiple
times is clearly not yielding the results you'd like.



Hi Matthew,
I'm testing WebRTC (JSSIP) with Asterisk 12.8 after following the
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support link.
Using Firefox, I can connect both JSSIP Clients to asterisk. When I Call
one Client, the Client just Ring One Time and after pick up a receive
WebRTC error on the Firefox browser.
Here is my asterisk sip debug:

<--- SIP read from WS:192.168.2.103:49851 ---> INVITE
sip:6000@192.168.2.106 SIP/2.0
Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK8394689
Max-Forwards: 69
To: 
From: "6001" ;tag=m6bqn333dr
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6407 INVITE
X-Can-Renegotiate: false
Contact: 
Content-Type: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: ice,replaces,outbound
User-Agent: JsSIP 2.0.2
Content-Length: 3158

v=0
o=mozilla...THIS_IS_SDPARTA-47.0.1 5760840281459352758 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256
E8:B7:2A:C2:DF:8B:AA:74:E6:D6:93:1C:68:88:81:39:82:C2:31:45:3D:8C:23:DF:
C1:23:72:03:F6:61:CC:F6
a=group:BUNDLE sdparta_0 sdparta_1
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 56808 UDP/TLS/RTP/SAVPF 109 9 0 8 c=IN IP4 87.169.189.102
a=candidate:0 1 UDP 2122187007 2003:88:6908:7659:3dbc:5101:33ff:d07a
56806 typ host
a=candidate:2 1 UDP 2122121471 2003:88:6908:7659:6113:7316:1ebc:43fa
56807 typ host
a=candidate:4 1 UDP 2122055935 192.168.2.103 56808 typ host
a=candidate:6 1 UDP 2122252543 192.168.56.1 56809 typ host a=candidate:0
2 UDP 2122187006 2003:88:6908:7659:3dbc:5101:33ff:d07a
56810 typ host
a=candidate:2 2 UDP 2122121470 2003:88:6908:7659:6113:7316:1ebc:43fa
56811 typ host
a=candidate:4 2 UDP 2122055934 192.168.2.103 56812 typ host
a=candidate:6 2 UDP 2122252542 192.168.56.1 56813 typ host
a=candidate:5 1 UDP 1685856255 87.169.189.102 56808 typ srflx raddr
192.168.2.103 rport 56808
a=candidate:5 2 UDP 1685856254 87.169.189.102 56812 typ srflx raddr
192.168.2.103 rport 56812
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=fmtp:109 maxplaybackrate=48000;stereo=1 a=ice-
pwd:138f583004cb3079134e8e8f20dac36f
a=ice-ufrag:0941ac54
a=mid:sdparta_0
a=msid:{fb724d76-44fe-4e7d-a8d8-e4c00b4b57fe}
{bba6da45-42c8-4529-8f4b-046cffcdc40d}
a=rtcp:56812 IN IP4 87.169.189.102
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=setup:actpass
a=ssrc:540714091 cname:{ecde75c0-993f-44af-b136-8944915fe31c}
m=video 56816 UDP/TLS/RTP/SAVPF 120 126 97 c=IN IP4 87.169.189.102
a=candidate:0 1 UDP 2122187007 2003:88:6908:7659:3dbc:5101:33ff:d07a
56814 typ host
a=candidate:2 1 UDP 2122121471 2003:88:6908:7659:6113:7316:1ebc:43fa
56815 typ host
a=candidate:4 1 UDP 2122055935 192.168.2.103 56816 typ host
a=candidate:6 1 UDP 2122252543 192.168.56.1 60290 typ host a=candidate:0
2 UDP 2122187006 2003:88:6908:7659:3dbc:5101:33ff:d07a
60291 typ host
a=candidate:2 2 

Re: [asterisk-users] PJSIP Multipart Body

[Simon Hohberg]

On 06/27/2016 12:09 PM, Joshua Colp wrote:

Simon Hohberg wrote:

Hi,

I want to pass a part of a SIP INVITE multipart body. I found a quite
old patch here:
https://issues.asterisk.org/jira/browse/ASTERISK-14510?jql=text%20~%20%22body%20part%22


But this patch is for the SIP channel driver not PJSIP, right?

Is it even possible without a patch? What do I have to put in the
dialplan then?


If you are asking if you can manipulate or get this information from the
dialplan in PJSIP it's not currently possible.



Hi Joshua,

thank you for taking time to come back to me.

It would be enough to just pass this body part on to the callee.

What about the SIP channel driver, is there a way to do this?


Regards,

Simon

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[asterisk-users] PJSIP Multipart Body

[Simon Hohberg]

Hi,

I want to pass a part of a SIP INVITE multipart body. I found a quite 
old patch here: 
https://issues.asterisk.org/jira/browse/ASTERISK-14510?jql=text%20~%20%22body%20part%22

But this patch is for the SIP channel driver not PJSIP, right?

Is it even possible without a patch? What do I have to put in the 
dialplan then?



Thanks in advance,

Simon

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Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

[Simon Hohberg]

Is it implied here that both HTTPS and WSS must also come from the 
same server (Same Origin Policy) ?

No, the same origin policy does not apply to web sockets.

Then, can I also install my own WebRTC demo page on my own private  
Asterisk server and access this demo page through HTTPS ?

If I'm not mistaken, this should fulfill all requirements.
It doesn't matter where the asterisk server is hosted. It is important 
where the web application comes from. If you don't want to use https and 
wss you only have the option to host the web app locally (on the same 
machine as the browser that loads the page), which probably makes sense 
only for development. Otherwise you have to use https and wss for the 
reasons discussed earlier.


Hope it helps.


Simon

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Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

[Simon Hohberg]

Hi Oliver,

On 02/18/2016 12:10 PM, Olivier wrote:

Hello,

I'm trying to have my first calls with WebRTC.
My server has asterisk 13.7.0.

I'm following the instructions from the wiki [1].
So I'm using [2] live demo from a Chrome navigator (v48) on Debian 
Jessie station.


Whenever I type something like ws://123.123.123.123:8088/ws 
 in Expert Mode form (see [1]), I'm 
getting this error :
*2:SecurityError: Failed to construct 'WebSocket': An insecure 
WebSocket connection may not be initiated from a page loaded over HTTPS.*
If I replace ws://123.123.123.123:8088/ws 
 with wss://123.123.123.123:8088/ws 
, this error message becomes with

/Disconnected:*Failed to connet to the server*/

My questions are:
1. Is wss now required by sipml5 live demo (implying wiki page is not 
up-to-date) ?
Yes, like the error says, you have to use wss on pages served via https. 
Furthermore, Chrome requires the use of https when you want to use 
getUserMedia.
See here: 
https://developers.google.com/web/updates/2015/10/chrome-47-webrtc?hl=en. It 
says: " Starting with Chrome 47, getUserMedia() requests are only 
allowed from secure origins: HTTPS or localhost."


The solution for development is, to host the webrtc client locally, so 
that you load the page from localhost. In that case getUserMedia is 
allowed with http, too (as the quote says). That means you have to 
download the dubango client and run a webserver on your dev machine.



2. Do you have any pointer for WebRTC with Asterisk 13 and PJSIP ?
Unfortunately, there is not much documentation about this, as far as I 
can tell.




Regards

[1] 
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5

[2] https://www.doubango.org/sipml5/





Regards,

Simon
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[asterisk-users] Delayed start of video with WebRTC - Missed FIR due to DTLS?

[Simon Hohberg]

Hi,

I am using Asterisk 13.7.0 with PJSIP.

I set up Asterisk for use with WebRTC SIP clients. After I managed to 
get video working, I noticed, that the calling party receives no video 
till 90s (or so) have passed. After 90s both parties receive video 
perfectly.


I am suspecting that this is due to the time needed for the DTLS 
handshake between Asterisk and the caller. Since Asterisk first 
establishes a full connection to the callee, the callee already begins 
sending data, while Asterisk is still performing the DTLS handshake with 
the caller. As a consequence the caller misses the first RTCP Full 
Intraframe Request (FIR) and the received video stream cannot be 
rendered till the next FIR 90s later arrives.


Am I right or is this nonsense?
Is this a known issue? I couldn't find anything about this.
Is there a fix available?


Thanks in advance!

Simon

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