SipML5 to softphone via webrtc2sip and asterisk-realtime

[Charles Colbourn]

Hi,

I've made a little progress. I can now set up a call from the softphone to the browser via webrtc2sip and asterisk. The reverse direction fails with an error though, and it seems to depend on a softphone that supports srtp.

The setup is as follows:

Browser (firefox) -> webrtc2sip ->asterisk (13.3.2) ->SFLPhone on Ubuntu

SFLPhone is set up with SRTP key exchange via SDES, and 'fallback on RTP on SDES failure'.

If I place a call from the FF browser to the SFLPhone the call is placed, but then dropped when the call is answered at the remote end. A warning appears on the asterisk console:

[May  6 16:46:44] WARNING[20191][C-0000000d]: chan_sip.c:10484 process_sdp: Failing due to no acceptable offer found

(I've attached SipML5 log, webrtc2sip log, plus the asterisk peers entries for both extensions [60002 is the browser, 60007 is the SFLPhone], plus the dialplan and sip.conf).

I've got two problems here

1) I haven't been able to place a call to any other softphone (Linphone, Ekiga) at all. Only SFLPhone gets as far as setting up the call, and that seems to be because it supports SDES

2) I can only place a call from SFLPhone to the browser, not from the browser back to SFLPhone.

I think what's happening is the configuration wants the connection to be encrypted from end to end - from the softphone to the browser. I need to be able to support SIP softphones that don't have encryption though. I understand that the webrtc standards require the browser->webrtc2sip connection to be encrypted, but can't the connection from Asterisk to the softphone be left unencrypted?

Could someone suggest where I'm going wrong?

Thanks,

Charles.

[Mamadou]

Asterisk is returning 503 [1] when webrtc2sip sends the INVITE. This is not a bug in sipml5 or webrtc2sip.

[1]

RECV:SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.241:10060;branch=z9hG4bK-249831501;received=192.168.1.241;rport=10060
From: <sip:60...@192.168.1.241>;tag=1948622290
To: <sip:60...@192.168.1.241>;tag=as0b20a28d
Call-ID: 3607aaa4-ffeb-babd-99ef-574167523657
CSeq: 465132839 INVITE
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Bearer capability not available
X-Asterisk-HangupCauseCode: 58
Content-Length: 0

[Charles Colbourn]

no, it's a bug in my ability to set the whole suite up successfully. I just hoped it would be obvious to someone who knew the software better where I've got it wrong. I'll try the asterisk mailing lists on monday.

[Charles Colbourn]

I cross posted this to the asterisk forums on May 12th, and in spite of hundreds of views I've had no response at all :-(

http://forums.asterisk.org/viewtopic.php?f=1&t=94896&sid=8f92f5fdcb62397a8f9d0d335b413ca8

I think I'm going to have to rip out the realtime stuff/build up a separate example server and try to replicate someone elses working webrtc2sip->asterisk setup, unless anyone has any suggestions/better ideas. That's going to be a bit time consuming though.

[navaismo]

I have used webrtc2sip + asterisk with realtime but different versions without problems. Maybe you need to re check your configs

[Charles Colbourn]

I've finally got to the bottom of this - it's a quirk in asterisk realtime config

sip.conf

encryption=true

db.sip_peers

encryption=false for the sip phone

encryption=true for the sipml5 phone

doesn't seem to work.

sip.conf

encryption=false for the sip phone

db.sip_peers

encryption=false for the sip phone

encryption=true for the sipml5 phone

does work. So the sip.conf global setting appears to be overriding the realtime setting where encryption is set to true globally in the sip.conf. Now that I describe it, I can see how that would be a desirable feature!

On Friday, 8 May 2015 23:56:47 UTC+1, Charles Colbourn wrote: