asterisk sip.conf tls encryption

0
votes

I am stuck in sometime with asterisk encryption.

sip.conf reload without any problem, as dial-plan, registering sip clients - no problem at all

When I call form one zoiper sip account to another wireshark capture tcp eth traffic shows following lines:

192.168.13.252    192.168.13.253    RTP    224    PT=ITU-T G.711 PCMU, SSRC=0x4C8C7A63, Seq=2259, Time=3154311440
192.168.13.253    192.168.13.252    SKYPE    224    Audio Unk: 5
192.168.13.253    192.168.13.252    SKYPE    224    Audio Unk: 5
192.168.13.253    192.168.13.252    SKYPE    224    Audio Unk: 5
192.168.13.252    192.168.13.253    RTP    224    PT=ITU-T G.711 PCMU, SSRC=0x4C8C7A63, Seq=2260, Time=3154311600
192.168.13.252    192.168.13.253    RTP    224    PT=ITU-T G.711 PCMU, SSRC=0x4C8C7A63, Seq=2261, Time=3154311760
192.168.13.253    192.168.13.252    SKYPE    224    Audio Unk: 5 ...

192.168.13.253 - asterisk server

192.168.13.252 - android phone (zoiper)

The problem is no sound on both phones during phone calls. Both phones send packages but not receiving any.

That is the SKYPE protocol involved in it? It suppose to be all line of RTP protocol.

2
encryptionasterisk

2 Answers

0
votes

If you are registering through SIP but receiving no audio, then for some reason your higher ports used for RTP are not receiving the data, most likely. Usually these ports are 10000-20000. Make sure both IPs can talk to each other through ports 5060-5061 and the higher ports. Can you display your asterisk CLI output while trying to make a call?

asterisk -vvvvvvvvvvvr

0
votes

Great let's get some details. rtp.conf 

[general]
rtpstart=10000
rtpend=20000

No errors reloading sip. This is interesting:

####CLI ### asterisk -vvvvvvvvvvvr #### shows
 == Using SIP RTP CoS mark 5
       > 0x7fb264004c00 -- Strict RTP learning after remote address set to: 192.168.13.104:58136
    -- Executing [200@phones:1] Dial("SIP/201-0000000b", "SIP/200") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/200
    -- SIP/200-0000000c is ringing
       > 0x7fb2440062f0 -- Strict RTP learning after remote address set to: 192.168.13.106:62856
    -- SIP/200-0000000c answered SIP/201-0000000b
    -- Channel SIP/200-0000000c joined 'simple_bridge' basic-bridge <9726e2bc-f161-452c-b489-c1829af2ed70>
    -- Channel SIP/201-0000000b joined 'simple_bridge' basic-bridge <9726e2bc-f161-452c-b489-c1829af2ed70>
       > 0x7fb264004c00 -- Strict RTP switching to RTP target address 192.168.13.104:58136 as source
       > 0x7fb2440062f0 -- Strict RTP switching to RTP target address 192.168.13.106:62856 as source
       > 0x7fb264004c00 -- Strict RTP learning complete - Locking on source address 192.168.13.104:58136
       > 0x7fb2440062f0 -- Strict RTP learning complete - Locking on source address 192.168.13.106:62856
    -- Channel SIP/201-0000000b left 'simple_bridge' basic-bridge <9726e2bc-f161-452c-b489-c1829af2ed70>
    -- Channel SIP/200-0000000c left 'simple_bridge' basic-bridge <9726e2bc-f161-452c-b489-c1829af2ed70>
  == Spawn extension (phones, 200, 1) exited non-zero on 'SIP/201-0000000b'
####

According to the CLI console information everything is in order. Asterisk runs on local IP, no firewall.