[Spce-user] No audio between subscribers

Alex Lutay alutay at sipwise.com
Mon Aug 16 04:10:37 EDT 2021


Dear Edo,

Please use one thread to discuss your issue,
at the moment you have created 3 non-related threads
> https://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/2021-August/thread.html

The RTF files you have shared is not the base way to share SIP dumps.
Please consider to use tool "sngrep" where you can filter
for the specific call and export it as tcpdump binary file.
At the moment RTF files contain a lot of unnecessary OPTIONs and 
REGISTER SIP messages and barely readable.

Anyway from the RTF files, NGCP should use primary G722 codec for the call:

A => NGCP:

c=IN IP4 192.168.1.17
m=audio 4010 RTP/AVP 9 0 8 3 102 120 101
a=rtpmap:9 G722/8000


B => NGCP:

o=- 0 1 IN IP4 192.168.0.250
m=audio 4008 RTP/AVP 9 101
a=rtpmap:9 G722/8000


NGCP => A:

c=IN IP4 127.0.0.1
m=audio 30140 RTP/AVP 9 101
a=rtpmap:9 G722/8000

For some reason RTPengine set 127.0.0.1 as a NGCP IP,
are you sure sip_ext network type is properly assigned in network.yml 
config?

See more information in NGCP documentation:

https://www.sipwise.com/doc/mr9.5.1/spce/ce/mr9.5.1/network-config/network-config.html

As you can see, here rtpengine is talking to himself on localhost:

> Aug  9 04:23:16 spce rtpengine[831]: INFO: [«tsg5T28gIKNxM6drC3HUlw..»]: [core] --------- Port       127.0.0.1:30176 <> «      127.0.0.1:8001 », SSRC «0», 0 p, 0 b, 0 e, 60 ts
> 
> Aug  9 04:23:16 spce rtpengine[831]: INFO: [«tsg5T28gIKNxM6drC3HUlw..»]: [core] --------- Port       127.0.0.1:30177 <> «      127.0.0.1:8002 » (RTCP), SSRC «0», 0 p, 0 b, 0 e, 60 ts

I hope it helps. Enjoy NGCP!

On 8/9/21 6:48 AM, [ EXT ] Edo wrote:
> All,
> 
> I have configured the system  (Linux spce 5.10.0-8-amd64 #1 SMP Debian 
> 5.10.46-2 (2021-07-20) x86_64) and attempting to make calls between 2 
> subscribers.
> Call is successful but no audio is available on both sides. Log does not 
> show any major anomaly except saying
> 
> -- Media #1 (audio over RTP/AVP) using unknown codec over SSRC and RTCP
> 
> Please help
> Thanks
...

-- 
Alex Lutay




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