ios DTLS-SRTP constraint causes thread assertion crash in Debug.

[Yide Pu]

Hi

  I 'm build r8005 and r7817 run lib on ios8.1, using a slightly modified version of the excellent pristine script:

https://github.com/pristineio/webrtc-build-scripts

crash in srtpfilter.cc srtp_init() 

bool SrtpSession::Init() {

  if (!inited_) {

    int err;

    err = srtp_init();

    if (err != err_status_ok) {

      LOG(LS_ERROR) << "Failed to init SRTP, err=" << err;

      return false;

    }

this my log：

Jingle:Net[en0:192.168.1.0/24:Unknown]: Allocation Phase=SslTcp

VAdapt Frame: scaled 0 / out 270 / in 270 Changes: 0 Input: 480x640 i33333333 Scale: 1 Output: 480x640 i33333333 Changed: false

Jingle:Conn[0x172a7360:audio:EyDERsff:1:0:local:tcp:192.168.1.114:62109->:1:1518149375:local:tcp:192.168.1.113:51652|C-WS|6520401919104138750|-]: Received STUN ping response eIuRVANt8ERe, pings_since_last_response_=89073444 89073545 89073649 89073752 89073856 89073961 89074070 89074177 89074281 89074393 89074499 89074607 89074709 89074813 89074915 , rtt=5

Jingle:Channel[audio|1|R_]: New best connection: Conn[0x170a61f0:audio:WjyDdowH:1:0:local:udp:192.168.1.114:49463->:1:2122129151:local:udp:192.168.1.113:64112|CRWI|9114475305677503998|-]

Jingle:Conn[0x170a61f0:audio:WjyDdowH:1:0:local:udp:192.168.1.114:49463->:1:2122129151:local:udp:192.168.1.113:64112|CRWS|9114475305677503998|-]: Received STUN ping response fYs/cNuf5Pzi, pings_since_last_response_=89073344 89073395 89073492 89073599 89073701 89073801 89073907 89074019 89074124 89074229 89074340 89074445 89074551 89074657 89074759 89074862 89074970 89075071 , rtt=138

BeginSSL: with peer

BeginSSL: as client

ContinueSSL

Would have blocked

Timeout is 1000 ms

Jingle:Channel[audio|1|__]: DtlsTransportChannelWrapper: Started DTLS handshake

Jingle:Conn[0x172a7360:audio:kx12UbBn:1:0:prflx:tcp:192.168.1.114:62110->:1:1518149375:local:tcp:192.168.1.113:51652|C-xS|6520401919775227390|1710]: Connection deleted due to read or write timeout

Jingle:Channel[audio|1|RW]: Removed connection (1 remaining)

All candidates gathered for audio:1:0

Transport: audio, component 1 allocation complete

Transport: audio allocation complete

Candidate gathering is complete.

NSSStreamAdapter::OnEvent SE_READ

ContinueSSL

NSSStreamAdapter::AuthCertificateHook

Checking against specified digest

Accepted peer certificate

Client cert requested

Would have blocked

Timeout is 1000 ms

NSSStreamAdapter::OnEvent SE_READ

ContinueSSL

Would have blocked

Timeout is 999 ms

NSSStreamAdapter::OnEvent SE_READ

ContinueSSL

Handshake complete

Jingle:Channel[audio|1|__]: DTLS handshake complete.

SetNetworkTransmissionState: 1

Channel socket writable (audio, 1) for the first time

webrtc: (vie_network_impl.cc:56): virtual void webrtc::ViENetworkImpl::SetNetworkTransmissionState(const int, const bool): channel: 0 transmitting: yes

Using Cand[27784895:1:udp:2122129151:192.168.1.114:49463:local::0:S6sh+gXOBquvoCiB:JXTtEQ80rKFiWuh3hUs5nbnJ]->Cand[2624638470:1:udp:2122129151:192.168.1.113:64112:local::0:W+Frgfw+eyWvJNYt:KbQ0Jhsjm+82mfqpaae9bEsY]

Installing keys from DTLS-SRTP on audio RTP

testing rand_source...

ICEConnectionStateChanged===Connected

passed

cipher: aes cipher block chaining (cbc) mode

  instance count: 0

  self-test: passed

cipher: aes integer counter mode

  instance count: 0

  self-test: passed

cipher: null cipher

  instance count: 0

  self-test: passed

auth func: hmac sha-1 authentication function

  instance count: 0

  self-test: passed

auth func: null authentication function

  instance count: 0

  self-test: passed

debug modules loaded:

  srtp (off)

  hmac sha-1 (off)

  aes cbc (off)

  aes icm (off)

  alloc (off)

  stat test (off)

  cipher (off)

  auth func (off)

  crypto kernel (off)

Error(srtpfilter.cc:706): Failed to init SRTP, err=2

Warning(channel.cc:853): DTLS-SRTP key installation failed

Error(common.cc:59): ../../webrtc/base/thread.cc(347): ASSERT FAILED: !current || current->blocking_calls_allowed_ @ AssertBlockingIsAllowedOnCurrentThread