Softphones don't work for outgoing class if they share extensions with WEBRTC phone. "488 not acceptable" error

FreePBX Endpoints
1

PBX: FreePBX 15.0.16.76
sip driver: PJSIP is configured for everything
by default we use custom WEBRTC phones and Zulu Desktop so that every extension has the following settings set:
Enable AVPF -> Yes
Enable ICE Support ->; Yes
Enable rtcp Mux -> Yes
Media Encryption -> DTLS-SRTP
Enable DTLS -> Yes
WEBRTC calls works perfectly.

But now we want to use normal sip software phone with the same extension like Microsip, phonerlite, jitso, etc. Tried many of them with different settings (security enabled/disabled/mandatory/optional etc , SRTP/ZRY/ etc), but was not able to make calls from these softphones, answering/receiving call s also works perfectly.

SUCCESFUL incoming call:04:57:41,005: T: 10.3.10.36:5060 (UDP)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.10.36:5060;rport=5060;branch=z9hG4bKPjfcc62d45-b357-4238-bbd4-b81749a415e2
From: “VY” sip:[email protected];tag=cb98d450-7cfd-4aa2-b18c-2cf2657782ad
To: sip:[email protected];tag=00e81cab281ceb11a519f15ad7a12669
Call-ID: 3ab18a1b-62c2-4883-b7ee-cdd245dbdce9
CSeq: 4385 INVITE
Contact: sip:[email protected]:5060;gr=00B69161-B21B-EB11-A25F-F15AD7A12669
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Session-Expires: 1800;refresher=uac
Supported: 100rel, replaces, from-change, gruu, timer
Server: PhonerLite 2.84
Content-Length: 314

v=0
o=- 2436188229 1 IN IP4 10.3.13.93
s=PhonerLite 2.84
c=IN IP4 10.3.13.93
t=0 0
m=audio 5062 UDP/TLS/RTP/SAVPF 107 9 8 0 18
a=rtpmap:107 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtcp-mux
a=ssrc:3401235981
a=sendrecv

04:57:41,007: R: 10.3.10.36:5060 (UDP)
ACK sip:[email protected]:5060;gr=00B69161-B21B-EB11-A25F-F15AD7A12669 SIP/2.0
Via: SIP/2.0/UDP 10.3.10.36:5060;rport;branch=z9hG4bKPj4a8b8d7b-87b4-4d12-81e9-ae7c3749abf4
From: “VY” sip:[email protected];tag=cb98d450-7cfd-4aa2-b18c-2cf2657782ad
To: sip:[email protected];tag=00e81cab281ceb11a519f15ad7a12669
Call-ID: 3ab18a1b-62c2-4883-b7ee-cdd245dbdce9
CSeq: 4385 ACK
Max-Forwards: 70
User-Agent: FPBX-15.0.16.76(15.7.3)
Content-Length: 0

UNSUCCESFULL outgoing call:

05:15:15,259: T: 10.3.10.36:5060 (UDP)
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.3.13.93:5060;branch=z9hG4bK807b53222b1ceb11a52bf15ad7a12669;rport
From: “PhonerLite” sip:[email protected];tag=3050302757
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 342 INVITE
Contact: sip:[email protected]:5060;gr=00B69161-B21B-EB11-A25F-F15AD7A12669
Authorization: Digest username=“145”, realm=“asterisk”, nonce=“XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX”, uri="sip:[email protected]", response=“XXXXXXXXXXXXXXXXXXXXXX”, algorithm=MD5, cnonce=“XXXXXXXXXXXXXXXXXXXXXX”, opaque=“XXXX”, qop=auth, nc=00000001
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
User-Agent: PhonerLite 2.84
Session-Expires: 900;refresher=uac
Supported: 100rel, replaces, from-change, gruu, timer
P-Preferred-Identity: sip:[email protected]
Content-Length: 400

v=0
o=- 2832770025 1 IN IP4 10.3.13.93
s=PhonerLite 2.84
c=IN IP4 10.3.13.93
t=0 0
m=audio 5062 RTP/SAVP 107 8 0 9 18
a=rtpmap:107 opus/48000/2
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:dj8165F+opL030HpHH1phK6gaNe5nOBxLnxsFJjr
a=encryption:optional
a=ssrc:3738291275
a=sendrecv

05:15:15,260: R: 10.3.10.36:5060 (UDP)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.3.13.93:5060;rport=5060;received=10.3.13.93;branch=z9hG4bK807b53222b1ceb11a52bf15ad7a12669
Call-ID: [email protected]
From: “PhonerLite” sip:[email protected];tag=3050302757
To: sip:[email protected]
CSeq: 342 INVITE
Server: FPBX-15.0.16.76(15.7.3)
Content-Length: 0

05:15:15,260: R: 10.3.10.36:5060 (UDP)
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.3.13.93:5060;rport=5060;received=10.3.13.93;branch=z9hG4bK807b53222b1ceb11a52bf15ad7a12669
Call-ID: [email protected]
From: “PhonerLite” sip:[email protected];tag=3050302757
To: sip:[email protected];tag=dce15ea1-47e5-41f5-83e4-0278833d338f
CSeq: 342 INVITE
Server: FPBX-15.0.16.76(15.7.3)
Content-Length: 0

I have tried many options both on software pheon site and extension confoguration side and was not able to allow to work both WEBRTC and soft phone together.

Please help!
i saw couple of similar topics on the foum , but they were unanswered and not finished, unfortunately.

2

You can’t use the same one for both WebRTC and a normal device, the media configuration is different and incompatible between the two.

3

so why incoming calls are successful and can be accepted by both WEBRTC phone and standard sip phone ? only outgoing calls from softphone don’t work with 488 error.

4

A community contributor tweaked the behavior for incoming to allow it to work, but noone has as of yet added support for doing it on outbound. The problem is that an outbound offer of WebRTC is incompatible with what the endpoint is expecting so it is rejected. Additional code and logic has to be added to support this. On incoming you can determine whether an offer is WebRTC or not, and adjust accordingly.

1 Like
5

Thanks a lot , Joshua! i created working solution with 2 extensions per user where secondary extension has “mailbox” and “CID Num Alias” modifed ( identical to first extension) .

6

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