Anybody know how to correct this?
My S20 drops audio on reinvite when making outbound calls.
[SDP]
v=0
o=- 327374591 327374591 IN IP4 192.168.60.9
s=Asterisk
>>>>> c=IN IP4 192.168.60.9 //this private IP is resulting in dropped audio (RTP stream) at 15min mark. This must be corrected.
t=0 0
m=audio 10478 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
There are a number of settings, but it would help to know the scenario. You presumably have a phone which is connected to the local LAN as is the PBX, the phone makes an external call which is processed out to the dialed number, but is this to a SIP provider, a remote phone, linkus?
Handsets are Yealink T29G's (about 6)
LAN is 192.168.60.x 255.255.255.0
Yeastar S20 with the IP address of 192.168.60.9.
ISP is Optimum Business w/ a static IP
Firewall is a Netgear ProSafe FVS318Gv2 with the IP address of 192.168.60.1 and I have NAT enabled.
I have SIP registration trunks from two different providers
1 is trunk is from VOIPVOIP that I use for inbound calls and the other is from FLOWROUTE that we use for outbound calls.
Yeastar configuration
Settings>System>Network>Mode = Single
LAN IP 192.168.60.9/255.255.255.0
Gateway 192.168.60.1
DNS 8.8.8.8 & 8.8.4.4
Settings>System>General>SIP>NAT
NAT Type = Disable
NAT Mode = Yes
Everything works fine until I make a call for more than 15 minutes out the FLOWROUTE route. The call doesn’t drop but we lose audio.
Testing:
When I configure Settings>System>General>SIP>NAT as External host
External IP address = Firewalls external address:5060
Local network 192.168.60.9/255.255.255.0
NAT Mode Yes (or No)
I can make outbound calls through the FLOWROUTE route without loss of audio. Issue seemed to be resolved. Problem is that my VOIPVOIP Trunk then fails to register.
When I try and set Settings>System>General>SIP>NAT back to Disable the VOIPVOIP trunk still fails to register. The only way I can get the system to function again is to restore from backup that was taken just before making the changes.
This is what FLOWROUTE is telling me:
Here is the INVITE we're getting from your side:
2020-02-26 14:43:49 -0500 : 24.89.180.26:5060 -> 10.200.2.51:5060
INVITE sip:1201988XXXX@sip.flowroute.com:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.60.9:5060;rport;branch=z9hG4bKPje7c60ab8-8eb3-407a-b9cb-9abfbfad3a38
From: "GRENIC" <sip:1-848-202-XXXX@sip.flowroute.com>;tag=4bea6378-18f5-4fd0-9194-6c507286ca9a
To: <sip:1201988XXXX@sip.flowroute.com>
>>>>> Contact: <sip:1-848-202-XXXX@192.168.60.9:5060>. //would be good if you can advertise your Public IP here as well, but this is not a root cause calls are dropping
Call-ID: b6a5cd76-d4c7-4bb7-83ad-4be9e9ac5b7e
CSeq: 2378 INVITE
Route: <sip:us-west-or.sip.flowroute.com:5060;transport=udp;lr>
Allow: OPTIONS, NOTIFY, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, REFER, REGISTER, MESSAGE
Supported: timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "1-848-202-XXXX" <sip:1-848-202-XXXX@sip.flowroute.com>
Max-Forwards: 70
User-Agent: Yeastar S20-30.12.0.9
Proxy-Authorization: Digest username="03626613", realm="sip.flowroute.com", nonce="XlbLoV5WynVobxfbRhjFmcRq2q6rs97c", uri="sip:1201988XXXX@sip.flowroute.com:5060;transport=udp", response="c08cc90aae49469fbc7d61ad05d3c618", cnonce="9479d208-3ed6-4d02-b8e3-1b30bdeaf4fd", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length: 306
[SDP]
v=0
o=- 327374591 327374591 IN IP4 192.168.60.9
s=Asterisk
>>>>> c=IN IP4 192.168.60.9 //this private IP is resulting in dropped audio (RTP stream) at 15min mark. This must be corrected.
t=0 0
m=audio 10478 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
Please review your PBX configuration as well as routed (possible SIP ALG settings) to define where you're advertising your private IP instead of public one
Hi There,
You could try to refer the solution here: https://support.yeastar.com/hc/en-us/articles/360023344094?page=1#comment_360004226353
1. disable SIP ALG in the router if not already
2. The NAT page should reflect (but using your IP and ports as I have a number of different devices behind the NAT and am not able to use 5060 on this particular PBX).
In your description of the test conditions, you indicated you had set NAT type to Disable. This is in error. You need to tell the PBX what the expectation is of it when responding. By having it set to disable, it did not know what it should use in the response and as a result, it simply used the only IP it knew, its own private IP. You want to change the NAT type to external IP.