No audio after attended transfer with pure SIP phone after M62 release

[Vasiliy Ganchev]

Hi, community!

my environment: 

- SIP server kamailio+rtpengine+asterisk

- pure SIP hardware phones (no ICE/DTLS support)

- WebRTC phones (JSSIP based)

After the release of latest Chrome M62 I have following issue (only audio call):

1. User-A (pure SIP) calls => User-B (WebRTC)

2. User-B sets call on hold (to make an attended transfer)

3. User-B makes new call   => User-C (WebRTC)

4. User-B finishes the transfer (in Asterisk there is a bridge of channels)

5. Asterisk sends re-INVITE to User-C with updated SDP 

after this - Chrome of User-C does not play audio, in logs I see (repeated every several seconds):

...

WARNING:statscollector.cc(1015)] The SSRC xxxxxxxxxx is not associated with a receiving track

...

WARNING:packet_buffer.cc(97)] Packet buffer flushed

...

The main difference between 2 INVITEs (on steps 3. and 5.) is:

for "3." user-C received SDP with "a=SSRC..." lines,

for "5." user-C received SDP without them

The same scheme worked with Chrome M61 (and I compared the logs, M61 ~ M62: they are pretty similar (except continuous buffer flushing warnings))

If I make some "hacks", to prevent sending re-INVITE from step "5." - audio is OK (but in logs I still see "The SSRC xxxxxxxxxx is not associated with a receiving track" warning)

Any ideas/suggestions would be highly appreciated (I would like to understand what changed in Chrome - to plan what is the preferred way to fix the issue.)

Thanks in advance!

BR, Vasiliy