[Solved] No audio in outgoing calls until the call is placed on-hold and off-hold

Asterisk Asterisk SIP
1

Hi everyone,

I am doing some bespoke development work with Asterisk and I’m stuck on this issue.

When you make an outgoing call there is no audio until it is placed on-hold and off-hold.

For incoming calls, the audio seems to work fine (from another SIP Provider to a UK number with Zadarma).

I have tested using Groundwire on iPhone and X-Lite on PC with the same result. The call flow is a phone in LAN, to VULTR server in WAN, to Zadarma SIP Provider, to a UK number with another SIP Provider (Yay).

I set up an extension on the Asterisk server which just plays music and that audio always seems to work fine. It’s only when calling out over Zadarma that I see this issue so I don’t know where the issue is but I think it is further upstream.

I tried adding some settings for NAT (even though the server has its own IPv4 and isn’t using NAT). I also tried adding some settings to the endpoint such as directmedia=no in attempt to fix the issue.

I am using the default rtp.conf sample file.

Here is my pjsip.conf file with passwords removed and SIP user ID changed:

; --- --- --- Start of File "pjsip.conf"

; --- --- --- Start of Transports

; --- Start of Transport "transport-udp"

[transport-udp]
type=transport
protocol=udp
bind=45.76.185.48
; Configuring res_pjsip to work through nat: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT
local_net=127.0.0.1/32
external_media_address=45.76.185.48
external_signaling_address=45.76.185.48

; --- End of Transport "transport-udp"

; --- --- --- End of Transports

; --- --- --- Start of Trunks

; --- Start of Trunk "zadarma"

[54321]
type=registration
transport=transport-udp
outbound_auth=54321
server_uri=sip:sip.zadarma.com
client_uri=sip:54321@sip.zadarma.com
retry_interval=60
expiration=120
contact_user=54321

[54321]
type=auth
auth_type=userpass
password=
username=54321

[54321]
type=aor
contact=sip:sip.zadarma.com:5060

[54321]
type=endpoint
transport=transport-udp
context=LocalSets
allow=!all,ulaw
outbound_auth=54321
aors=54321
from_user=54321
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
 
[54321]
type=identify
endpoint=54321
match=sip.zadarma.com
match=185.45.152.0/24
match=185.45.155.0/24
match=37.139.38.0/24
match=195.122.19.0/27

; --- End of Trunk "zadarma"

; --- --- --- End of Trunks

; --- --- --- Start of Templates

; --- Start of Template "endpoint"

[endpoint](!)
type=endpoint
transport=transport-udp
context=internal
allow=!all,ulaw
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
 
[auth-userpass](!)
type=auth
auth_type=userpass

[aor-single-reg](!)
type=aor
max_contacts=10

; --- End of Template "endpoint"

; --- --- --- End of Templates

; --- --- --- Start of Endpoints

; --- Start of Endpoint "1001"

[1001](endpoint)
auth=1001
aors=1001

[1001](auth-userpass)
password=
username=1001

[1001](aor-single-reg)

; --- End of Endpoint "1001"

; --- Start of Endpoint "1002"

[1002](endpoint)
auth=1002
aors=1002

[1002](auth-userpass)
password=
username=1002

[1002](aor-single-reg)

; --- End of Endpoint "1002"

; --- --- --- End of Endpoints

; --- --- --- End of File "pjsip.conf"
1 Like
2

What version of Asterisk are you using? I remember something like this in the issue tracker but I believe it was resolved.

1 Like
3

How about capturing a call and check if any ports are changed when placing the call on hold and continue the call.

I would start there.

1 Like
4

Here’s the output of core show version:

Asterisk certified/13.18-cert3 built by root @ virtual-receptionist on a x86_64 running Linux on 2018-02-26 17:24:24 UTC
5

Then I’d do as @meightee stated and also provide the “pjsip set logger on” output and “rtp set debug on” to show the actual traffic (both SIP and media).

6

Here we go (I’ve replaced the SIP User ID with 54321, the caller’s IP with 1.2.3.4 and telephone numbers with 442081234567 and 442031234567):

Gist: https://gist.github.com/anonymous/ecd3b6b8026604aedef88d4602112815

Raw: https://gist.githubusercontent.com/anonymous/ecd3b6b8026604aedef88d4602112815/raw/83562064da3aea85ab082f5ca252e3c5a8316b31/Asterisk%20console%20output

7

There is no media being received on either side before the call is put on hold. Is Asterisk really not behind some kind of 1 to 1 NAT or other router?

You might try enabling rtp_keepalive which will cause Asterisk to send RTP keepalive packets instead of relying on the other side.

2 Likes
8

Added rtp_keepalive=1 to the Zadarma endpoint and the audio works!

2 Likes