One-way audio issue with PJSIP: assincronous codecs

[mbello]

Hi, following this discussion on FreePBX forums:

http://community.freepbx.org/t/fpbx13-rc...l/31042/51

It appears that Yealink phones are the culprit.

I captured pcap files that I believe will help understand the problem. The pcap is for a call from a Yealink T46g phone to an asterisk 13.5 box, the call is to the Echo service (*43).

The call should go like this:
- First, Asterisk plays a message that explains how the echo service works. This message lasts for about 25 seconds;
- Then the echo test begins and everything I say is echoed back to me.

Looking at the pcap, it happened like this:
- Asterisk on SDP offered codecs ALAW and G722;
- Yealink T46G offered G722 and ALAW;
- Asterisk started sendng G711 but then 1 second after, it switched to G722 to play the message. I could not hear anything that was played, the phone was silent;
- During this time while asterisk was sending audio in G722 format, the T46g phone was sending back in G711 format;
- After the initial announcement played, Asterisk went back to sending audio in codec G711 and I then could here the echo test.

Other notes:
- If on Asterisk I make G722 the first option than everything works fine.;
- With exactly the same codec priority as set on the T46g, my Bria Android client worked just fine and I could hear the announcement;
- However, grandstream phones also suffered the same problem, I could not hear the announcement from echo test.

Could Yealink please investigate this issue?