# Intermittent one way voice with Asterisk 13, 1.4 work no issue **URL:** \ **Category:** Asterisk SIP\ **Created:** [November 10, 2018, 6:22am UTC](https://community.asterisk.org/t/intermittent-one-way-voice-with-asterisk-13-1-4-work-no-issue/77076 "2018-11-10T06:22:00Z")\ **Posts on this page:** 4\ **Page:** 1
**Author:** ![bobp](https://avatars.discourse-cdn.com/v4/letter/b/eb9ed0/32.png) [@bobp](https://community.asterisk.org/u/bobp)\ **Post date:** [November 10, 2018, 6:22am UTC](https://community.asterisk.org/t/intermittent-one-way-voice-with-asterisk-13-1-4-work-no-issue/77076/1 "2018-11-10T06:22:00Z")
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? ---
**Author:** ![poing](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/poing/32/1696_2.png) [@poing](https://community.asterisk.org/u/poing)\ **Post date:** [November 13, 2018, 6:14am UTC](https://community.asterisk.org/t/intermittent-one-way-voice-with-asterisk-13-1-4-work-no-issue/77076/2 "2018-11-13T06:14:44Z")
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. ---
**Author:** ![bobp](https://avatars.discourse-cdn.com/v4/letter/b/eb9ed0/32.png) [@bobp](https://community.asterisk.org/u/bobp)\ **Post date:** [November 20, 2018, 10:13pm UTC](https://community.asterisk.org/t/intermittent-one-way-voice-with-asterisk-13-1-4-work-no-issue/77076/3 "2018-11-20T22:13:46Z")
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)… ---
**Author:** ![poing](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/poing/32/1696_2.png) [@poing](https://community.asterisk.org/u/poing)\ **Post date:** [November 21, 2018, 3:34am UTC](https://community.asterisk.org/t/intermittent-one-way-voice-with-asterisk-13-1-4-work-no-issue/77076/4 "2018-11-21T03:34:08Z")
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. ```auto 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: ```auto 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" ;tag=as5ac0fc73 To: Contact: 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" ;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" ;tag=as5ac0fc73 To: ;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: 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 ```