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[djchillerz at gmail.com]

I have two accounts on Asterisk 13. One uses chan_sip and the other pjsip.


In my snom 760 the setup for these two accounts is identical.


When I call echo test from the account using chan_sip audio comes through fine.


When I call echo test from the account using pjsip there is no audio.


With rtp set debug on, I can see that audio is being sent to the snom's internal IP 192.168.0.x


I can add a stun server in the config for this account and RTP flows to the Public IP and I get audio.


I was wondering why there is a difference between pjsip and chan_sip so that one works without stun and the other requires it.  Does anybody know why? Maybe my settings are off in pjsip.


Here's how I have my endpoint configured:


[test]
type=endpoint
context=dial_out
disallow=all
allow=alaw
allow=speex
allow=speex16
allow=speex32
allow=gsm
allow=ulaw
allow=g722
auth=test
aors=test
direct_media=no
media_encryption=sdes
media_encryption_optimistic=yes
rtp_symmetric=yes
force_rport=no
rewrite_contact=yes  ; necessary if endpoint does not know/register public ip:port
ice_support=yes    ;This is specific to clients that support NAT traversal
                   ;for media via ICE,STUN,TURN. See the wiki at:
                   ;https://wiki.asterisk.org/wiki/x/D4FHAQ
                   ;for a deeper explanation of this topic.

[test]
type=auth
auth_type=userpass
password=redacted
username=test

[test]
type=aor
remove_existing=yes
max_contacts=2
qualify_frequency=60


Looking forward to your thoughts.


Kind Regards,


C

[jcolp at digium.com]

Chirag Desai wrote:

[djchillerz at gmail.com]

[jcolp at digium.com]

Chirag Desai wrote:

[djchillerz at gmail.com]

[jcolp at digium.com]

Chirag Desai wrote:

[djchillerz at gmail.com]

I'm dialling from the snom and every few calls asterisk sends media to the phones external IP and it works!


And then now and again it sends the media to the phones internal IP and I hear nothing. I'm really at a loss.

[jcolp at digium.com]

Chirag Desai wrote:

[djchillerz at gmail.com]

In the PCAP I can see asterisk sending UDP packets to my local IP 192.168.0.5....


It's funny, when I switch to TCP on 5060 audio seems to work fine. The moment I go to 5063 on TLS everything goes a bit awry. Any further input is greatly appreciated.

[jcolp at digium.com]

Chirag Desai wrote:

[djchillerz at gmail.com]

So I see:


EXTERNAL_SNOM_IP -> EXTERNAL_ASTERISK_IP   (UDP,  length 218, src: 60798, dst 11128)


EXTERNAL_ASTERISK_IP -> INTERNAL_SNOM_IP   (UDP, length 218, src: 11128 dst 60478


So i see udp from the phone, but there's no audio.



I do also see some packets ::


EXTERNAL_ASTERISK_IP -> EXTERNAL_SNOM_IP (ICMP, length 246, Destination unreachable (Host administratively prohibited)

[jcolp at digium.com]

Chirag Desai wrote:

[djchillerz at gmail.com]

You were right. I had non-default rtp ports open in iptables. Edited rtp.conf et voila. Everything seems to be working.


Thanks so much for your patience and guidance!


Have a lovely eening.

[pete at fiberphone.co.nz]