Using a webRTC endpoint in linphone or zoiper

Asterisk Asterisk Endpoints
1

Heya, here is a question.

Currently we are using jsSIP for web calls in the browser which is working great. Using the following endpoint configuration:

[6B64CAE6-C721-4F47-B2BD-2CE8EA00332B]
type=endpoint
context=chattabai
allow=!all,alaw,ulaw,opus
rtp_timeout=30
aors=6B64CAE6-C721-4F47-B2BD-2CE8EA00332B
auth=6B64CAE6-C721-4F47-B2BD-2CE8EA00332B
webrtc=yes

[6B64CAE6-C721-4F47-B2BD-2CE8EA00332B]
type=aor
max_contacts=1
remove_existing=yes

[6B64CAE6-C721-4F47-B2BD-2CE8EA00332B]
type=auth
auth_type=userpass
username=A9511596-58C8-4524-AB73-CA425C26B215
password=26DD401E-8303-4D06-A197-FE48175E681D

Now we also want to use this same endpoint with a soft phone app like zoiper or linphone. When using this endpoint in a soft phone app the registration works. But when a caller is calling the callee we get the response “Call result: CHANUNAVAIL”

 Executing [1@ctbVariableOutboundCallStartSuccess:10] Set("PJSIP/6B64CAE6-C721-4F47-B2BD-2CE8EA00332B-0000001f", "ctbEndPointEndpoint=6B64CAE6-C721-4F47-B2BD-2CE8EA00332B") in new stack
    -- Executing [1@ctbVariableOutboundCallStartSuccess:11] Set("PJSIP/6B64CAE6-C721-4F47-B2BD-2CE8EA00332B-0000001f", "_ctbVisitorUserId=2800") in new stack
    -- Executing [1@ctbVariableOutboundCallStartSuccess:12] Dial("PJSIP/6B64CAE6-C721-4F47-B2BD-2CE8EA00332B-0000001f", "PJSIP/6B64CAE6-C721-4F47-B2BD-2CE8EA00332B,30,b(ctbVariableOutboundCallStartHeaders^addheader^1)") in new stack
    -- PJSIP/6B64CAE6-C721-4F47-B2BD-2CE8EA00332B-00000020 Internal Gosub(ctbVariableOutboundCallStartHeaders,addheader,1) start
    -- Executing [addheader@ctbVariableOutboundCallStartHeaders:1] Set("PJSIP/6B64CAE6-C721-4F47-B2BD-2CE8EA00332B-00000020", "PJSIP_HEADER(add,CtbVisitorUserId)=2800") in new stack
    -- Executing [addheader@ctbVariableOutboundCallStartHeaders:2] Return("PJSIP/6B64CAE6-C721-4F47-B2BD-2CE8EA00332B-00000020", "") in new stack
  == Spawn extension (chattabai, 1, 1) exited non-zero on 'PJSIP/6B64CAE6-C721-4F47-B2BD-2CE8EA00332B-00000020'
    -- PJSIP/6B64CAE6-C721-4F47-B2BD-2CE8EA00332B-00000020 Internal Gosub(ctbVariableOutboundCallStartHeaders,addheader,1) complete GOSUB_RETVAL=
    -- Called PJSIP/6B64CAE6-C721-4F47-B2BD-2CE8EA00332B
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1@ctbVariableOutboundCallStartSuccess:13] Verbose("PJSIP/6B64CAE6-C721-4F47-B2BD-2CE8EA00332B-0000001f", "Call result: CHANUNAVAIL") in new stack
Call result: CHANUNAVAIL

When using the following endpoint configuration in zoiper or line phone everything works. But not for webRTC calls

[local1]
type=endpoint
transport=transportSecure
context=chattabai
allow=!all,ulaw,alaw
aors=local1
media_encryption=sdes
rtp_symmetric=yes
;direct_media=no
;disable_direct_media_on_nat=yes
;force_rport=yes
;ice_support=yes
dtmf_mode=rfc4733
rewrite_contact=yes

[local1]
type=aor
max_contacts=1
remove_existing=yes

[local1]
type=auth
auth_type=userpass
password=local1
username=local1

So the question is, are you able to create 1 endpoint that is both available (compatible) for webRTC and soft phone calls like zoiper and line phone. Or do I need to create 2 separate endpoints. One for the webRTC calls and one for the soft phone calls?

Regards Jerome

2

You have to create two separate endpoints. WebRTC SDP is incompatible with normal endpoints, unless they support it.

1 Like
3

Thank you for the quick response, just found more information about this here Configuring Asterisk for WebRTC Clients - Asterisk Documentation

Have some reading todo :wink:

Regards Jerome

4

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