Gnome-calls audio on Fedora 40

Asterisk Asterisk Endpoints
1

Hi,

I am really trying to get the Gnome SIP client working; gnome-calls.

I know it’s an audio issue as you will see, below.

Any help would sincerely be appreciated!

gnome-call to vm with audio fail:

<--- Received SIP request (1018 bytes) from UDP:192.168.x.x-g-c-client:51919 --->
INVITE sip:vmext@asteriskbox:5060 SIP/2.0
Via: SIP/2.0/UDP gc-clientIP:51919;rport;branch=z9hG4bK4mKZKK342mQFN
Max-Forwards: 70
From: <sip:myext@asteriskbox>;tag=NDr37QSH5g3DB
To: <sip:vmext@asteriskbox:5060>
Call-ID: 124c674f-836b-123e-3594-64006a80530e
CSeq: 96862035 INVITE
Contact: <sip:myext@asteriskbox>
User-Agent: calls sofia-sip/1.13.16
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE
Supported: timer, 100rel, replaces, gruu, outbound
Authorization: Digest username="myext", realm="asterisk", nonce="1742831652/c94d31bfab32c707f0ee38e561ea7ad1", cnonce="EkydJYNrEj6UNWQAaoBTDg", opaque="7cd2131819b32bb4", algorithm=MD5, uri="sip:vmext@asteriskbox:5060", response="915bdb77c68ae8d10da4ffda55ebf0ec", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 146

v=0
o=- 1568758603753376869 2465131287975409939 IN IP4 gc-clientIP
s=-
c=IN IP4 gc-clientIP
t=0 0
m=audio 0 RTP/AVP 9 8 0 3
a=inactive

<--- Transmitting SIP response (320 bytes) to UDP:gc-clientIP:51919 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP gc-clientIP:51919;rport=51919;received=gc-clientIP;branch=z9hG4bK4mKZKK342mQFN
Call-ID: 124c674f-836b-123e-3594-64006a80530e
From: <sip:myext@asteriskbox>;tag=NDr37QSH5g3DB
To: <sip:vmext@asteriskbox>
CSeq: 96862035 INVITE
Server: Asterisk PBX 18.4.0
Content-Length:  0

gnome-calls log details (using pastbin to reduce length of this post):
https://pastebin.com/6ZqL1Fyy

2

Asterisk log is incomplete.

However the call is started, by the client, in a held state, which is not normally a good thing.

Could not create pipeline: Could not retrieve used socket from RTP udpsrc

I’m no familiar with the client’s logs, but I suspect that this is a consequence of port 0, which which is one of the indications of hold, rather than the cause.

3

My apologies! I cot overconfident and thought I knew what I was doing. Thank you for your help!

This should be complete:

<--- Received SIP request (632 bytes) from UDP:gc-clientIP:37571 --->
REGISTER sip:asteriskbox:5060 SIP/2.0
Via: SIP/2.0/UDP gc-clientIP:37571;rport;branch=z9hG4bKU08ZN5X94552F
Max-Forwards: 70
From: <sip:myext@asteriskbox>;tag=Ugj2QQ6H1F8Dj
To: <sip:myext@asteriskbox>
Call-ID: 33b17b5b-8393-123e-73a9-64006a80530e
CSeq: 96870652 REGISTER
Contact: <sip:myext@gc-clientIP:37571;user=phone>;methods="INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,MESSAGE,SUBSCRIBE,NOTIFY,REFER,UPDATE"
Expires: 180
User-Agent: calls sofia-sip/1.13.16
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE
Supported: timer, 100rel, replaces, gruu, outbound
Content-Length: 0


<--- Transmitting SIP response (499 bytes) to UDP:gc-clientIP:37571 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP gc-clientIP:37571;rport=37571;received=gc-clientIP;branch=z9hG4bKU08ZN5X94552F
Call-ID: 33b17b5b-8393-123e-73a9-64006a80530e
From: <sip:myext@asteriskbox>;tag=Ugj2QQ6H1F8Dj
To: <sip:myext@asteriskbox>;tag=z9hG4bKU08ZN5X94552F
CSeq: 96870652 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1742848888/557da1e967588d2a2b6b47c1db3785b3",opaque="101464ff40579fca",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.4.0
Content-Length:  0


<--- Received SIP request (914 bytes) from UDP:gc-clientIP:37571 --->
REGISTER sip:asteriskbox:5060 SIP/2.0
Via: SIP/2.0/UDP gc-clientIP:37571;rport;branch=z9hG4bKv91rQ0eD2evNB
Max-Forwards: 70
From: <sip:myext@asteriskbox>;tag=Ugj2QQ6H1F8Dj
To: <sip:myext@asteriskbox>
Call-ID: 33b17b5b-8393-123e-73a9-64006a80530e
CSeq: 96870653 REGISTER
Contact: <sip:myext@gc-clientIP:37571;user=phone>;methods="INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,MESSAGE,SUBSCRIBE,NOTIFY,REFER,UPDATE"
Expires: 180
User-Agent: calls sofia-sip/1.13.16
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE
Supported: timer, 100rel, replaces, gruu, outbound
Authorization: Digest username="myext", realm="asterisk", nonce="1742848888/557da1e967588d2a2b6b47c1db3785b3", cnonce="M7Gp0IOTEj6pc2QAaoBTDg", opaque="101464ff40579fca", algorithm=MD5, uri="sip:asteriskbox:5060", response="b3bebb90cbd04c37b39d17b299f2e4c4", qop=auth, nc=00000001
Content-Length: 0


    -- Added contact 'sip:myext@gc-clientIP:37571;user=phone' to AOR 'myext' with expiration of 180 seconds
<--- Transmitting SIP response (458 bytes) to UDP:gc-clientIP:37571 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP gc-clientIP:37571;rport=37571;received=gc-clientIP;branch=z9hG4bKv91rQ0eD2evNB
Call-ID: 33b17b5b-8393-123e-73a9-64006a80530e
From: <sip:myext@asteriskbox>;tag=Ugj2QQ6H1F8Dj
To: <sip:myext@asteriskbox>;tag=z9hG4bKv91rQ0eD2evNB
CSeq: 96870653 REGISTER
Date: Mon, 24 Mar 2025 20:41:28 GMT
Contact: <sip:myext@gc-clientIP:37571;user=phone>;expires=179
Expires: 180
Server: Asterisk PBX 18.4.0
Content-Length:  0


  == Endpoint myext is now Reachable
<--- Received SIP request (409 bytes) from UDP:gc-clientIP:37571 --->
OPTIONS sip:myext@asteriskbox SIP/2.0
v:SIP/2.0/UDP gc-clientIP:37571;rport;branch=z9hG4bKXjUHSUZgZQj8p
f:<sip:myext@asteriskbox>;tag=Ugj2QQ6H1F8Dj
t:<sip:myext@asteriskbox>
i:2Y3w-KKwaWwHarfgmwyuSf
CSeq:96870652 OPTIONS
a:*;require;explicit;methods="INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,MESSAGE,SUBSCRIBE,NOTIFY,REFER,UPDATE"
Accept:application/vnd.nokia-register-usage
s:REGISTRATION PROBE
l:0   


<--- Transmitting SIP response (484 bytes) to UDP:gc-clientIP:37571 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP gc-clientIP:37571;rport=37571;received=gc-clientIP;branch=z9hG4bKXjUHSUZgZQj8p
Call-ID: 2Y3w-KKwaWwHarfgmwyuSf
From: <sip:myext@asteriskbox>;tag=Ugj2QQ6H1F8Dj
To: <sip:myext@asteriskbox>;tag=z9hG4bKXjUHSUZgZQj8p
CSeq: 96870652 OPTIONS
WWW-Authenticate: Digest realm="asterisk",nonce="1742848888/557da1e967588d2a2b6b47c1db3785b3",opaque="1826dc423995d1b5",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.4.0
Content-Length:  0


<--- Received SIP request (681 bytes) from UDP:gc-clientIP:37571 --->
OPTIONS sip:myext@asteriskbox SIP/2.0
v:SIP/2.0/UDP gc-clientIP:37571;rport;branch=z9hG4bKyUmaUpgmv08tj
f:<sip:myext@asteriskbox>;tag=Ugj2QQ6H1F8Dj
t:<sip:myext@asteriskbox>
i:2Y3w-KKwaWwHarfgmwyuSf
CSeq:96870652 OPTIONS
a:*;require;explicit;methods="INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,MESSAGE,SUBSCRIBE,NOTIFY,REFER,UPDATE"
Accept:application/vnd.nokia-register-usage
Authorization:Digest username="myext",realm="asterisk",nonce="1742848888/557da1e967588d2a2b6b47c1db3785b3",cnonce="M7cMl4OTEj6pc2QAaoBTDg",opaque="1826dc423995d1b5",algorithm=MD5,uri="sip:myext@asteriskbox",response="f60efc2845661058945f05fe6ac54bea",qop=auth,nc=00000001
s:REGISTRATION PROBE
l:0   


<--- Transmitting SIP response (813 bytes) to UDP:gc-clientIP:37571 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP gc-clientIP:37571;rport=37571;received=gc-clientIP;branch=z9hG4bKyUmaUpgmv08tj
Call-ID: 2Y3w-KKwaWwHarfgmwyuSf
From: <sip:myext@asteriskbox>;tag=Ugj2QQ6H1F8Dj
To: <sip:myext@asteriskbox>;tag=z9hG4bKyUmaUpgmv08tj
CSeq: 96870652 OPTIONS
Accept: application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/pidf+xml, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: Asterisk PBX 18.4.0
Content-Length:  0


<--- Transmitting SIP request (656 bytes) to UDP:gc-clientIP:37571 --->
NOTIFY sip:myext@gc-clientIP:37571;user=phone SIP/2.0
Via: SIP/2.0/UDP asteriskbox:5060;rport;branch=z9hG4bKPjff93fac8-0434-4750-9481-695981a1f6c7
From: <sip:myext@asteriskbox>;tag=8e307dc0-c11f-4807-984e-d71e52a82ad7
To: <sip:myext@gc-clientIP;user=phone>
Contact: <sip:myext@asteriskbox:5060>
Call-ID: 72bf2005-19f1-48c1-8513-261f7d09d28a
CSeq: 1428 NOTIFY
Subscription-State: terminated
Event: message-summary
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: Asterisk PBX 18.4.0
Content-Type: application/simple-message-summary
Content-Length:    49

Messages-Waiting: yes
Voice-Message: 2/1 (0/0)

<--- Received SIP response (540 bytes) from UDP:gc-clientIP:37571 --->
SIP/2.0 481 Subscription Does Not Exist
Via: SIP/2.0/UDP asteriskbox:5060;rport=5060;branch=z9hG4bKPjff93fac8-0434-4750-9481-695981a1f6c7
From: <sip:myext@asteriskbox>;tag=8e307dc0-c11f-4807-984e-d71e52a82ad7
To: <sip:myext@gc-clientIP;user=phone>;tag=vSBUSjQNyry0D
Call-ID: 72bf2005-19f1-48c1-8513-261f7d09d28a
CSeq: 1428 NOTIFY
User-Agent: calls sofia-sip/1.13.16
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE
Supported: timer, 100rel, replaces, gruu, outbound
Content-Length: 0


<--- Received SIP request (731 bytes) from UDP:gc-clientIP:37571 --->
INVITE sip:vmext@asteriskbox:5060 SIP/2.0
Via: SIP/2.0/UDP gc-clientIP:37571;rport;branch=z9hG4bKZ4D3vH1QS9yDe
Max-Forwards: 70
From: <sip:myext@asteriskbox>;tag=X24KUD8rU1mKS
To: <sip:vmext@asteriskbox:5060>
Call-ID: 3c192617-8393-123e-73a9-64006a80530e
CSeq: 96870659 INVITE
Contact: <sip:myext@asteriskbox>
User-Agent: calls sofia-sip/1.13.16
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE
Supported: timer, 100rel, replaces, gruu, outbound
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 146

v=0   
o=- 5569689873136906610 8123243188955567135 IN IP4 gc-clientIP
s=-   
c=IN IP4 gc-clientIP
t=0 0 
m=audio 0 RTP/AVP 9 8 0 3
a=inactive

<--- Transmitting SIP response (497 bytes) to UDP:gc-clientIP:37571 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP gc-clientIP:37571;rport=37571;received=gc-clientIP;branch=z9hG4bKZ4D3vH1QS9yDe
Call-ID: 3c192617-8393-123e-73a9-64006a80530e
From: <sip:myext@asteriskbox>;tag=X24KUD8rU1mKS
To: <sip:vmext@asteriskbox>;tag=z9hG4bKZ4D3vH1QS9yDe
CSeq: 96870659 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1742848902/84aa26b16f5fa7521979baa34a2cc0a4",opaque="71da6f5237c292c5",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.4.0
Content-Length:  0

<--- Received SIP request (321 bytes) from UDP:gc-clientIP:37571 --->
ACK sip:vmext@asteriskbox:5060 SIP/2.0
Via: SIP/2.0/UDP gc-clientIP:37571;rport;branch=z9hG4bKZ4D3vH1QS9yDe
Max-Forwards: 70
From: <sip:myext@asteriskbox>;tag=X24KUD8rU1mKS
To: <sip:vmext@asteriskbox>;tag=z9hG4bKZ4D3vH1QS9yDe
Call-ID: 3c192617-8393-123e-73a9-64006a80530e
CSeq: 96870659 ACK
Content-Length: 0


<--- Received SIP request (1018 bytes) from UDP:gc-clientIP:37571 --->
INVITE sip:vmext@asteriskbox:5060 SIP/2.0
Via: SIP/2.0/UDP gc-clientIP:37571;rport;branch=z9hG4bK0D7UycjUpjN0S
Max-Forwards: 70
From: <sip:myext@asteriskbox>;tag=X24KUD8rU1mKS
To: <sip:vmext@asteriskbox:5060>
Call-ID: 3c192617-8393-123e-73a9-64006a80530e
CSeq: 96870660 INVITE
Contact: <sip:myext@asteriskbox>
User-Agent: calls sofia-sip/1.13.16
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE
Supported: timer, 100rel, replaces, gruu, outbound
Authorization: Digest username="myext", realm="asterisk", nonce="1742848902/84aa26b16f5fa7521979baa34a2cc0a4", cnonce="PBlZ5YOTEj6pc2QAaoBTDg", opaque="71da6f5237c292c5", algorithm=MD5, uri="sip:vmext@asteriskbox:5060", response="094811be071b15c2ebf666d39ae50d56", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 146

v=0   
o=- 5569689873136906610 8123243188955567135 IN IP4 gc-clientIP
s=-   
c=IN IP4 gc-clientIP
t=0 0 
m=audio 0 RTP/AVP 9 8 0 3
a=inactive

<--- Transmitting SIP response (320 bytes) to UDP:gc-clientIP:37571 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP gc-clientIP:37571;rport=37571;received=gc-clientIP;branch=z9hG4bK0D7UycjUpjN0S
Call-ID: 3c192617-8393-123e-73a9-64006a80530e
From: <sip:myext@asteriskbox>;tag=X24KUD8rU1mKS
To: <sip:vmext@asteriskbox>
CSeq: 96870660 INVITE
Server: Asterisk PBX 18.4.0
Content-Length:  0


<--- Transmitting SIP response (374 bytes) to UDP:gc-clientIP:37571 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP gc-clientIP:37571;rport=37571;received=gc-clientIP;branch=z9hG4bK0D7UycjUpjN0S
Call-ID: 3c192617-8393-123e-73a9-64006a80530e
From: <sip:myext@asteriskbox>;tag=X24KUD8rU1mKS
To: <sip:vmext@asteriskbox>;tag=1720cb81-15e7-4966-a29d-52182f18759b
CSeq: 96870660 INVITE
Server: Asterisk PBX 18.4.0
Content-Length:  0


<--- Received SIP request (337 bytes) from UDP:gc-clientIP:37571 --->
ACK sip:vmext@asteriskbox:5060 SIP/2.0
Via: SIP/2.0/UDP gc-clientIP:37571;rport;branch=z9hG4bK0D7UycjUpjN0S

Max-Forwards: 70
From: <sip:myext@asteriskbox>;tag=X24KUD8rU1mKS
To: <sip:vmext@asteriskbox>;tag=1720cb81-15e7-4966-a29d-52182f18759b
Call-ID: 3c192617-8393-123e-73a9-64006a80530e
CSeq: 96870660 ACK
Content-Length: 0
4

It might be the initial held status, which is definitely not a normal thing to do, although I though it would be acceptable,

It could be that there are no codecs in common, but the logging isn’t sufficient to tell that, and you haven’t shown the endpoint configuration.

488 can also happen if Asterisk is set for secure RTP, or WebRTC, given this is SDP for standard, unencrypted, RTP.

5

The config. It could be codec-related as you postulated–I’m only allowing ulaw and g729. I’m going to search on codecs–I have no clue about them–do I install them on the client? OS? Somewhere else?

[1000]
type=endpoint
transport=transport-udp-nat
context=my-phone
;context=from-internal
disallow=all
allow=ulaw,g729
auth=1000_auth
aors=1000
callerid=MYNAME <##########>
;
; endpoint NAT required settings:
rtp_symmetric=no
force_rport=yes
rewrite_contact=yes
;
direct_media=yes
;dtmf_mode=rfc4733
;dtmf_mode=rfc2833
;
; MWI related options
aggregate_mwi=yes
mailboxes=1000@default
mwi_from_user=1000
;
; Extortion and Device state options
;
device_state_busy_at=1
allow_subscribe=yes
sub_min_expiry=30
;
; STIR/SHAKEN support.
;
;stir_shaken=no

[1000_auth]
type=auth
auth_type=userpass
password=mysecretpwIdidntchangeitdontlook
username=1000

[1000]
type=aor
remove_existing=yes
max_contacts=1
;contact=sip:1000@myclientIP:5060
6

ulaw should be accepted. I guess it doesn’t like the port 0 indication of an initial hold state. There are very few valid reasons to use G.729.

7

So ulaw alone is sufficient?
allow=ulaw ?

And I forgot to ask–can I provide more logging to confirm codec[s]? How do I know I have captured sufficient logging?

And last question: I looked at every “0” in the log and could not find the port you referenced as 0. Can you give me a pointer on “hold” and port 0 references?

Thank you very much!

8

When the log tells you why it sent 488.

The number after audio in the m= line is the port number to which to send media. That is set to 0. Setting it to 0 is the old way of signalling hold with SDP.

The a=inactive setting says that the stream is to started with no media in either direction. Anything except a=sendrecv there is the new way of sending a hold indication, except that the normal convention is to send a=sendlonly.

9

Asterisk requires a working audio stream in the initial call setup, thus because there isn’t one it sends a 488.

10

Actually, 0 in the m= is refusing the stream, which doesn’t even make sense, as this is an offer, not a response! I was thinking of c=0.0.0.0, but, half remembering, assumed the 0 in the m= must be a hold, as what is actually being sent makes no sense. The INVITE should not have been sent at all, as there was no way of honouring it!

11

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