Re: No audio in SIPML5 - Asterisk 11 - Chrome 23

[AutoStatic]

Are you using everything locally? And how are the two private networks connected? Did you try disabling RTCWeb Breaker in sipml5?

[Mamadou]

For sure what you're trying won't work: Using RTCWeb Breaker without
webrtc2sip and not patching Asterisk.
Please open a ticket in the tracker to get help
(http://code.google.com/p/sipml5/issues/entry).

On 12/27/2012 3:18 PM, Jos� Cervera wrote:
> Hi,
>
> After a couple of days trying, I can't manage to make the above setup
> work. I've never had success in getting the audio in Chrome.
> I've use the same Asterisk server between two softphones (XLite)
> without problems. I've tried chrome to Softphone, chrome to chrome,
> and finally, I'm just trying to have asterisk play a background sound.
>
> I'm quite new to SIP, so I'm not sure where I'm going wrong.
>
> sip.conf:
> -----------------------
>
> [general]
> context=default
> allowguest=no
> transport=ws,wss
> udpbindaddr=0.0.0.0:5060
>
> [1063]
> type=peer
> username=1063
> host=dynamic
> secret=1063
> context=demo
> hasiax=no
> hassip=yes
> encryption=yes
> avpf=yes
> icesupport=yes
> videosupport=no
> directmedia=no
> transport=ws
>
> extensions.conf:
> ----------------
> [demo]
> exten => 2602,1,Answer
> exten => 2602,2,Background(demo-instruct)
> exten => 2602,3,WaitExten()
> exten => 2602,4,Hangup
>
>
> In the Asterisk logs, if I enable RTP logs, it's as if Asterisk was
> trying to contact my IP via UDP:
> localhost*CLI> rtp set debug on
> RTP Debugging Enabled
> Sent RTP packet to 192.168.191.170:65284 (type 00, seq 003548, ts
> 068960, len 000170)
> Sent RTP packet to 192.168.191.170:65284 (type 00, seq 003549, ts
> 069120, len 000170)
> Sent RTP packet to 192.168.191.170:65284 (type 00, seq 003550, ts
> 069280, len 000170)
> [...]
>
> On the client, I don't see any incoming traffic with Wireshark. I've
> completely disabled the windows firewall, just in case.
> A tcpdump on the server on the port 65284 confirms that it is UDP.
> Isn't the media traffic supposed to go also in the websocket?
>
> You'll find attached the full Asterisk and console logs for the
> registration and the call attempt itself, in case you have a minute to
> check.
>
> Thanks and regards.
>
>
> --
>
>

[José Cervera]

Thank you both for your answers.

I misunderstood the instructions here: http://code.google.com/p/sipml5/wiki/Asterisk

I thought that RTCWeb Breaker was an alternative to patching Asterisk, I didn't get that webrtc2sip is needed in that situation.

I'll test tomorrow and let you know.

Regards

Le jeudi 27 décembre 2012 21:46:21 UTC+1, Mamadou a écrit :

For sure what you're trying won't work: Using RTCWeb Breaker without
webrtc2sip and not patching Asterisk.
Please open a ticket in the tracker to get help
(http://code.google.com/p/sipml5/issues/entry).

[Mamadou]

Make sense, not clearly said. Updated the wiki to say the webrtc2sip is required for RTCWeb Breaker.

--
 
 

[AutoStatic]

On Thursday, 27 December 2012 22:05:43 UTC+1, José Cervera wrote:

I thought that RTCWeb Breaker was an alternative to patching Asterisk, I didn't get that webrtc2sip is needed in that situation.

I'll test tomorrow and let you know.

Fwiw, I'm using a bog standard Asterisk 1.8.11 installation with sipml5 to do videocalls through my own webrtc2sip server. Works well and I'd recommend it over a patched Asterisk.

Jeremy

[Mamadou]

What's the address?

--
 
 

[AutoStatic]

On Thursday, 27 December 2012 22:54:29 UTC+1, Mamadou wrote:

What's the address?

Of the webrtc2sip server? Unfortunately I can't publish its fqdn or IP publicly for security reasons as it's a server of the company where I work. So it's actually not my own webrtc2sip server but one I build. Sorry for the wrong choice of words :(

[Mamadou]

Sorry I sent the message to the wrong thread :)
Off course I will not ask the address of your server.

--
 
 

[José Cervera]

Well, still no luck. I get to connect succesfully, but when I call, I have a "Forbidden":

using INVITE request as basis request - c33b4131-6110-e39b-6367-9d5dcfbd3d32

Found peer '1063' for '1063' from 172.22.1.27:10060

[Dec 28 12:55:36] NOTICE[30684][C-0000000a]: chan_sip.c:25008 handle_request_invite: Failed to authenticate device <sip:10...@172.22.1.27>;tag=1356737476322

However, the authentication is OK for the Register, as you can see in the attached logs.

[José Cervera]

I've hade problems sending the logs as .txt. Sending a single zip instead, sorry

[TJey]

I have same issue, did you resolve it ?

{{{

<--- SIP read from UDP:192.168.1.22:6060 --->
INVITE sip:1...@192.168.1.23 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.22:6060;branch=z9hG4bK1359322983046;rport
From: <sip:1...@192.168.1.23>;tag=1357253902879
To: <sip:1...@192.168.1.23>
Contact: <sip:1...@192.168.1.22:6060;ws-src-ip=192.168.1.24;ws-src-
port=49948;ws-src-proto=ws;transport=udp>
Call-ID: bcdacbab-3207-7546-8fbd-043122e2365f
CSeq: 707710850 INVITE
Content-Type: application/sdp
Content-Length: 1019
Max-Forwards: 70
Authorization: Digest
username="101",realm="asterisk",nonce="0288ad9d",uri="sip:
1...@192.168.1.23",response="6a7265ccadbe22039bf256c7c4f15c07",algorithm=MD5
User-Agent: webrtc2sip Media Server 2.0
P-Preferred-Identity: <sip:webrtc2sip>
Supported: 100rel

v=0
o=doubango 1983 678901 IN IP4 192.168.1.22
s=-
c=IN IP4 192.168.1.22
t=0 0
m=audio 36780 RTP/AVP 0 3 8 101
c=IN IP4 192.168.1.22
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=tcap:1 RTP/SAVPF
a=pcfg:1 t=1
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:
9c053KyubLazeAdJMjQfIbgVDBgHPyspXnZG2tcB
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:YIuFznFcpG/3eUR5COG1hDNKoc/
2WeC/Wznl5lOq
a=rtcp-mux
a=ssrc:1561808081 cname:ldjWoB60jbyQlR6e
a=ssrc:1561808081 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:1561808081 label:Doubango
a=ice-ufrag:Tyieomy3VNZVLrj
a=ice-pwd:hU2ZUYoCK47wfzXKuAijd
a=mid:audio
a=candidate:j5eZ9tNB4 1 udp 2130706431 192.168.1.22 36780 typ host
a=candidate:j5eZ9tNB4 2 udp 2130706430 192.168.1.22 36781 typ host
a=candidate:srflxj5eZ 2 udp 1694498814 178.88.3.165 15464 typ srflx
a=candidate:srflxj5eZ 1 udp 1694498815 178.88.3.165 15463 typ srflx
<------------->
--- (14 headers 30 lines) ---
Sending to 192.168.1.22:6060 (no NAT)
Using INVITE request as basis request -
bcdacbab-3207-7546-8fbd-043122e2365f
Found peer '100' for '101' from 192.168.1.22:6060

<--- Reliably Transmitting (no NAT) to 192.168.1.22:6060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
192.168.1.22:6060;branch=z9hG4bK1359322983046;received=192.168.1.22;rport=6060
From: <sip:1...@192.168.1.23>;tag=1357253902879
To: <sip:1...@192.168.1.23>;tag=as212f1a6f
Call-ID: bcdacbab-3207-7546-8fbd-043122e2365f
CSeq: 707710850 INVITE
Server: Asterisk PBX 11.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

}}}

On Dec 28 2012, 8:37 pm, José Cervera <jose.cerv...@gmail.com> wrote:
> Well, still no luck. I get to connect succesfully, but when I call, I have
> a "Forbidden":
>
> using INVITE request as basis request - c33b4131-6110-e39b-6367-9d5dcfbd3d32
> Found peer '1063' for '1063' from 172.22.1.27:10060
> [Dec 28 12:55:36] NOTICE[30684][C-0000000a]: chan_sip.c:25008
> handle_request_invite: Failed to authenticate device

> <sip:1...@172.22.1.27>;tag=1356737476322

[José Cervera]

Nope... Not really sure what's going on, I don't have enough experience to analyze it. If you find out, please come back to report what the problem is! 

Regards.

Le vendredi 4 janvier 2013 21:40:39 UTC+1, TJey a écrit :

I have same issue, did you resolve it ?

{{{

<--- SIP read from UDP:192.168.1.22:6060 --->

INVITE sip...@192.168.1.23 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.22:6060;branch=z9hG4bK1359322983046;rport

From: <sip...@192.168.1.23>;tag=1357253902879
To: <sip...@192.168.1.23>

From: <sip...@192.168.1.23>;tag=1357253902879
To: <sip...@192.168.1.23>;tag=as212f1a6f

[AutoStatic]

Please post your sip.conf files.

Regards,

Jeremy

[José Cervera]

I posted the relevant subset at the beginning of the thread. I don't have access to the full configuration now, but it was just other users' configuration, which shouldn't be important  (I guess) as I was just trying to play a wave. I can provide it next wednesday, though.

Thanks and regards,

José 

[AutoStatic]

Your transport settings are wrong, should be udp. You should use webrtc2sip as the WebSocket server, not Asterisk.

hasiax=no

hassip=yes

encryption=yes

avpf=yes

icesupport=yes

I think you can delete these settings, you don't need them, webrtc2sip takes care of this. Regarding the hasiax and hassip settings, I think these belong into the user.conf file, not sip.conf.

[TJey]

Hi Jose, i followed suggestions from guys Mamadou and AutoStatic -
thanks guys!

Which are :

1) Evalute correct Realm in when you trying to login ( my was just ip
address, but pc name was asterisk, i changed realm value to asterisk )
2) I removed these entries in sip.conf as long as i undertood, all
work is done in webrtc2sip and asterisk will think it is regular
extension )

hasiax = no
hassip = yes
encryption = yes
avpf = yes
icesupport = yes
videosupport=no
directmedia=no


p.s I am using
webrtc2sip and asterisk11 on two separate virtual machines

On Jan 6, 1:40 am, José Cervera <jose.cerv...@gmail.com> wrote:
> Nope... Not really sure what's going on, I don't have enough experience to
> analyze it. If you find out, please come back to report what the problem
> is!
> Regards.
>
> Le vendredi 4 janvier 2013 21:40:39 UTC+1, TJey a écrit :
>
>
>
>
>
>
>
>
>
> > I have same issue, did you resolve it ?
>
> > {{{
>
> > <--- SIP read from UDP:192.168.1.22:6060 --->

> > INVITE sip...@192.168.1.23 <javascript:> SIP/2.0

> > Via: SIP/2.0/UDP 192.168.1.22:6060;branch=z9hG4bK1359322983046;rport

> > From: <sip...@192.168.1.23 <javascript:>>;tag=1357253902879
> > To: <sip...@192.168.1.23 <javascript:>>

> > Contact: <sip:1...@192.168.1.22:6060;ws-src-ip=192.168.1.24;ws-src-
> > port=49948;ws-src-proto=ws;transport=udp>
> > Call-ID: bcdacbab-3207-7546-8fbd-043122e2365f
> > CSeq: 707710850 INVITE
> > Content-Type: application/sdp
> > Content-Length: 1019
> > Max-Forwards: 70
> > Authorization: Digest
> > username="101",realm="asterisk",nonce="0288ad9d",uri="sip:

> > 1...@192.168.1.23 <javascript:>",response="6a7265ccadbe22039bf256c7c4f15c07",algorithm=MD5

[José Cervera]

Hi,

thanks, I'll try that when I get back to work. I had already set the transport to udp when I installed webrtc2sip (forgot to mention that). However, I hadn't tried removing the other options. And as for the realm, I'm also using the IP, I'll try the hostname.

Thanks.

[TJey]

One thing is unclean for me, is
if I have 2 different machines webrtc2sip and asterisk separate
servers, what ip address/hostname should I specifiy in realm field ?
webrtc2sip's or asterisk's ip(realm)

[AutoStatic]

The realm of the Asterisk server.

[TJey]

Thanks for quick response, but i have no luck to run asterisk
+webrtc2sip+chrome23 :(

Please AutoStatic, could you answer these questions:

1) Why webrtc2sip quering DNS server, even if i am in private network
and have internal servers ( asterisk, webrtc2sip ), basicly IS
webrtc2sip only works when there is INTERNET connection ?

2) Could sombody post working configuration example for ASTERISK
+WEBRTC2SIP like here http://code.google.com/p/sipml5/wiki/Asterisk
sip.conf, extensions.conf, and webrtc2sip config.xml, i think it would
be helped to everybody.

p.s Now i have 488 Error NOT ACCEPTABLE HERE

[AutoStatic]

On Monday, 7 January 2013 15:44:37 UTC+1, TJey wrote:

Thanks for quick response, but i have no luck to run asterisk
+webrtc2sip+chrome23 :(

Please AutoStatic, could you answer these questions:

1) Why webrtc2sip quering DNS server, even if i am in private network
and have internal servers ( asterisk, webrtc2sip ), basicly IS
webrtc2sip only works when there is INTERNET connection ?

Try resolving hostnames within your private network, either by setting up a nameserver yourself or by adding the necessary hosts to your /etc/hosts file (Or Windows/system32/drivers/etc/hosts).

2) Could sombody post working configuration example for ASTERISK
+WEBRTC2SIP like here http://code.google.com/p/sipml5/wiki/Asterisk
sip.conf, extensions.conf, and webrtc2sip config.xml, i think it would
be helped to everybody.

I think your Asterisk and webrtc2sip configs are ok. Except for the contexts of the peers, I wouldn't set it to default unless you configured a dialplan for that context.
 

p.s Now i have 488 Error NOT ACCEPTABLE HERE

That's a codec issue. Make sure you're allowing at least PCMU and PCMA on both Asterisk and webrtc2sip. And I now see there's an error in your webrtc2sip config:
<codecs>pcma;pcmu;gsm;</codecs>

Should read:
<codecs>pcma;pcmu;gsm</codecs>