No voice over 3G/LTE VOIP/SIP

Asterisk Asterisk Support
1

Hi,

I have installed Asterisk on my dd-wrt router with a connection to my SIP provider upstream and I’m trying to get it working with two UAs.

One UA is connected on the home network over ethernet and in/out to this phone works great.

I am trying to connect to my Asterisk box from my Android phone using sipdroid over 4G and this does not work for making phone calls. The SIP signalling seems to be complete but there is no voice. I have captured the packets using Wireshark and I see the RTP traffic coming into the box but it’s not getting forwarded to Android phone from what I can tell. I seem to have a NAT type of problem even though both my Android phone and Asterisk server is on public networks from what I can tell.

I have tried playing with directmedia=yes/no; directrtpsetup=yes; nat=comedia and nothing seems to work.

Version is “Asterisk 1.8.22.0, Copyright © 1999 - 2012 Digium, Inc. and others.”

Here is my sip.conf

[general]
context=incoming
bindport=5060
bindaddr=0.0.0.0
recordhistory=yes
disallow=all
allow=ulaw
allow=gsm
trustrpid=yes
sendrpid=yes
dtmfmode=inband
relaxdtmf=yes
realm=xyz.dyndns.org
srvlookup=no
allowguest=no
directmedia=yes
;directmedia=no
;nat=comedia
;directrtpsetup=yes

externip=xyz.dyndns.org  ; EC2 instance IP address
localnet=192.168.1.1/255.255.255.0 ; local network as seen by .ifconfig eth0.
register=><mynumber>:<secret>@<mysip provider>


[1001]
type=friend
context=outbound
host=dynamic
secret=XXXX              
disallow=all
allow=gsm
allow=ilbc
outgoinglimit=1
incominglimit=1
;canreinvite=no

asterisk -rvvvvv output

Verbosity was 1 and is now 5
    -- Registered SIP '1001' at 72.22.182.10:52412
       > Saved useragent "Sipdroid/3.0 beta/SAMSUNG-SGH-I727" for peer 1001
    -- Registered SIP '1001' at 192.228.197.101:45307
    -- Unregistered SIP '1001'
    -- Registered SIP '1001' at 192.228.197.101:53208
  == Using SIP RTP CoS mark 5
    -- Executing [18003582227@outbound:1] Dial("SIP/1001-00000006", "SIP/18003582227@sipprovider,60,r") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/18003582227@sipprovider
    -- SIP/sipprovider-00000007 is making progress passing it to SIP/1001-00000006
    -- SIP/sipprovider-00000007 answered SIP/1001-00000006
[Jun 10 16:22:49] WARNING[2751]: dsp.c:1421 ast_dsp_process: Inband DTMF is not supported on codec gsm. Use RFC2833
  == Spawn extension (outbound, 18003582227, 1) exited non-zero on 'SIP/1001-00000006'

Your help is greatly appreciated
Thanks.

2

Can you please try doing a debug “sip set debug on” enabled?

3

hello:
please check this error: Inband DTMF is not supported on codec gsm. Use RFC2833