Intermittent one way voice with Asterisk 13, 1.4 work no issue

Environment: home office, with NAT, colo with 2 asterisk boxes (1.4 and 13.18.2) both use static public, standard SIP (not pjsip) with UDP

Trying to move from 1.4 to 13.18.2… Same phone provide configured on both. Almost identical configuration.
Phone registering to 1.4 - no issue
Same phone registering to Asterisk v13 - intermittent one way audio (call comes in, phone rings, people can hear me but I can’t hear them). Intermittent = out of 10 calls 1-2 would have an issue (can’t hear people but they can hear me). Only happens on incoming calls.

Differences in config
nat=yes on v.1.4 VS nat=force_rport,auto_comedia on v13
disallow=all (on both)
allow=ulaw (on both)
allow=gsm (on both)
canreinvite=no on 1.4 VS directmedia=no on v13
dtmfmode=rfc2833 on both

I was able to get wireshark trace of the problem call… Audio gets delivered to v13 but does not get delivered to the phone

I’m honestly lost… I have no clue what to look for…

Anyone has an advise?

Check your rtp port settings, for all devices involved.
One-way audio can be caused by a call-leg using an rtp port outside the allowed range.

1 Like

Right… but what am I looking for? start & end ports are set the same on version 1.4 and v13… range 10000-20000… I kind of understand the issue, just not understanding what to do… If it was NAT related, ok… that I can deal with… but I was getting the issue even with no NAT involved (public IP on the devices)…

You need to capture the SIP messages, using and application like wireshark -or- tcpdump at multiple points in the network path. It may take a few attempts to duplicate the issue.

You’ll want to find the sdp message. Each direction has an RTP port. The example below is using 19386 and 11374 µ-law between two (2) nodes in the network path.

IP 10.10.10.100.sip > 10.10.10.98.sip: UDP, length 910
 m=audio 19386 RTP/AVP 0 101
 a=rtpmap:0 PCMU/8000
IP 10.10.10.98.sip > 10.10.10.100.sip: UDP, length 752
 m=audio 11374 RTP/AVP 0 101
 a=rtpmap:0 PCMU/8000

Here’s a full tcpdump of the sdp between two nodes:

IP 10.10.10.100.sip > 10.10.10.98.sip: UDP, length 910
E...|\..@.....Fd..Fb........INVITE sip:5553334228@10.10.10.98 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK289178e2
Max-Forwards: 70
From: "4442221137" <sip:asterisk@asterisk.fqdn.tld>;tag=as5ac0fc73
To: <sip:5553334228@10.10.10.98>
Contact: <sip:asterisk@10.10.10.100:5060>
Call-ID: 0ff4e79e27c48cde4bfbcc15567710ce@asterisk.fqdn.tld
CSeq: 102 INVITE
User-Agent: asterisk
Date: Wed, 21 Nov 2018 01:22:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "4442221137" <sip:4442221137@asterisk.fqdn.tld>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 78687125 78687125 IN IP4 10.10.10.100
s=asterisk
c=IN IP4 10.10.10.100
t=0 0
m=audio 19386 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

IP 10.10.10.98.sip > 10.10.10.100.sip: UDP, length 752
E.......@..7..Fb..Fd........SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK289178e2;received=10.10.10.100
From: "4442221137" <sip:asterisk@asterisk.fqdn.tld>;tag=as5ac0fc73
To: <sip:5553334228@10.10.10.98>;tag=as05384e0b
Call-ID: 0ff4e79e27c48cde4bfbcc15567710ce@asterisk.fqdn.tld
CSeq: 102 INVITE
Server: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:5553334228@10.10.10.98:5060>
Content-Type: application/sdp
Content-Length: 220

v=0
o=root 2070816973 2070816973 IN IP4 10.10.10.98
s=asterisk
c=IN IP4 10.10.10.98
t=0 0
m=audio 11374 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv