Hi all,
I’ve run into some trouble with my Asterisk setup and I’m having trouble pin-pointing the exact cause.
I am making a call to asterisk using my chrome browser via websockets (SIPml) and am able to establish the call and audio flows fine. The trouble occurs when I hold and then resume the call. The resumed call has one way audio - the SIP phone can hear the Chrome user, but not the other way around.
Key
192.168.3.150 - Machine running Softphone on Chrome
192.168.3.100 - SIP phone endpoint
Below are the RTP packets BEFORE the call is held:
...
Sent RTP packet to 192.168.3.150:62115 (via ICE) (type 00, seq 004883, ts 326347392, len 4294967284)
Got RTP packet from 192.168.3.150:62115 (type 00, seq 020761, ts 3708907820, len 000160)
Sent RTP packet to 192.168.3.100:3000 (type 00, seq 047904, ts 3708907816, len 000160)
Got RTP packet from 192.168.3.100:3000 (type 00, seq 018764, ts 326347556, len 000160)
Sent RTP packet to 192.168.3.150:62115 (via ICE) (type 00, seq 004884, ts 326347552, len 4294967284)
Got RTP packet from 192.168.3.150:62115 (type 00, seq 020762, ts 3708907980, len 000160)
Sent RTP packet to 192.168.3.100:3000 (type 00, seq 047905, ts 3708907976, len 000160)
Got RTP packet from 192.168.3.100:3000 (type 00, seq 018765, ts 326347716, len 000160)
...
Here are the packets AFTER the resume:
...
Sent RTP packet to 192.168.3.150:62115 (via ICE) (type 00, seq 004888, ts 326366592, len 4294967284)
> 0x7f383c048330 -- Probation passed - setting RTP source address to 192.168.3.150:62895
Got RTP packet from 192.168.3.150:62895 (type 00, seq 002373, ts 3708927018, len 000160)
Sent RTP packet to 192.168.3.100:3000 (type 00, seq 048021, ts 3708927016, len 000160)
Got RTP packet from 192.168.3.100:3000 (type 00, seq 018884, ts 326366756, len 000160)
Sent RTP packet to 192.168.3.150:62115 (via ICE) (type 00, seq 004889, ts 326366752, len 4294967284)
Got RTP packet from 192.168.3.150:62895 (type 00, seq 002374, ts 3708927178, len 000160)
Sent RTP packet to 192.168.3.100:3000 (type 00, seq 048022, ts 3708927176, len 000160)
Got RTP packet from 192.168.3.100:3000 (type 00, seq 018885, ts 326366916, len 000160)
...
As is evident by wireshark packets, the packets being sent to port 62115 are all failing as the port number has changed to 62895.
I’m wondering why the port has changed, and why Asterisk isn’t trying to use this new port to send the packets to.
Any help is greatly appreciated.
Thank you.
Kind Regards,
Gary Shergill