PJSIP - IPv4 Trunk and IPv6 client - RTP is not forwarded

Asterisk Asterisk SIP
1

Asterisk with 2 network interfaces, LAN (IPv4) and WAN (IPv6).

LAN side has a trusted local IPv4 peer/trunk (No NAT), and I’m trying to get remote IPv6 clients working. Clients can register, and calls can be made (from client, outbound via trunk), but no audio is heard.

I have verified we are receiving RTP from either leg of the call, but Asterisk does not forward the RTP on to the other leg of the call.

sip.conf:


[trunk]
username=trunk
secret=pass
type=friend
host=x.x.x.x
nat=no
insecure=port,invite
dtmfmode=rfc2833
context=from-trunk
canreinvite=yes
disallow=all
allow=alaw
allow=ulaw
allow=g729

[2000]
type=friend
host=dynamic
secret=pass
dtmfmode=rfc2833
context=from-2000
directmedia=no
disallow=all
allow=alaw
allow=ulaw
allow=g729

When the call is made, log output is:

    -- Registered SIP '2000' at [2001:8004::1234]:25158
  == Using SIP RTP CoS mark 5
       > 0x7f8c9808f3b0 -- Strict RTP learning after remote address set to: [2001:8004::1234]:20000
    -- Executing [01234567@from-2000:1] Dial("SIP/2000-0000001a", "SIP/trunk-2000/01234567,60") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/trunk-2000/01234567
    -- SIP/trunk-2000-0000001b is ringing
       > 0x7f8c9c00f230 -- Strict RTP learning after remote address set to: 10.0.0.1:8216
    -- SIP/trunk-2000-0000001b is making progress passing it to SIP/2000-0000001a
    -- SIP/trunk-2000-0000001b answered SIP/2000-0000001a
    -- Channel SIP/trunk-2000-0000001b joined 'simple_bridge' basic-bridge <b129ca16-3efc-402d-829e-a26ce3511b0b>
    -- Channel SIP/2000-0000001a joined 'simple_bridge' basic-bridge <b129ca16-3efc-402d-829e-a26ce3511b0b>
       > Bridge b129ca16-3efc-402d-829e-a26ce3511b0b: switching from simple_bridge technology to native_rtp
       > Locally RTP bridged 'SIP/2000-0000001a' and 'SIP/trunk-2000-0000001b' in stack
    -- Channel SIP/2000-0000001a left 'native_rtp' basic-bridge <b129ca16-3efc-402d-829e-a26ce3511b0b>
  == Spawn extension (from-2000, 01234567, 1) exited non-zero on 'SIP/2000-0000001a'
    -- Channel SIP/trunk-2000-0000001b left 'native_rtp' basic-bridge <b129ca16-3efc-402d-829e-a26ce3511b0b>

I’m at a loss on what to do, any tips from people who have got this working would be apprecaited.

2

You are using an unsupported channel driver, and have not provided any RTP debugging or SIP protocol trace.

3

Hi David,

I’ve changed to PJSIP, and it has the same issue.

What debug command would you want enabled?

4

pjsip set logger on
rtp set debug on
core set verbose 5
core set debug 5
uncomment the full log in logger.conf.

5

OneDrive Link of debug.7z. Expires in 48 hours.

Contains:
/var/log/asterisk/full
/etc/asterisk/pjsip.conf
/etc/asterisk/rtp.conf
(a tcpdump of the last call)

NOTE: Skip to the last call in the debug log. Previous attempts had some auth issues.

User experience:

  • Extension (2319) registers and tries to attempt call, rings briefly, answered, but no-way audio.
    We can see from the trace RTP is delivered from either leg source, but not forwarded to appropriate dest.
6

Hi David,

Have you had a chance to look through the logs yet?

-L

7

I don’t know if I can even decode 7Z. Please look at them yourself, and extract the relevant INVITE transactions and typical RTP debug lines for both directions for both legs, noting any missing ones.

8

.7z is just an archive, like zip.

7-zip.org - Free, open source compression tool (much like WinZIP back in the day).

9

I’m not being paid for this, so you should be minimising the work I have to do, by making it as easy as possible to access files, and by analysing and pruning them as much as possible, yourself.

10

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