no audio with srtp/rtp_forward=no

[abma]

when i disable rtp_forward then i have no audio. it seems yate receives the rtp stream but doesn't forward it to the other leg.

i.e. phone a sends rtp to yate, yate receives the rtp packet but doesn't forward it to phone b

an idea what could cause this?

with rtp_forward=yes it works.

Code: [Select]

<INFO> Could not classify call from '303', wasted 9 usec
<INFO> Could not route call to '304' in context 'default', wasted 950 usec
<cdrbuild:INFO> Got message 'call.route' for untracked id 'sip/89'
<yrtp:ALL> RTP/SAVP message received
<yrtp:ALL> YRTPWrapper::YRTPWrapper('<yate server ip>',0x7f98fc02e760,'audio',bidir,0x7f98fc02acc0,false) [0x7f98fc0419a0]
<yrtp:ALL> YRTPWrapper::setupRTP("<yate server ip>",true,true) [0x7f98fc0419a0]
<yrtp:INFO> Session 'yrtp/305743342' 0x7f98fc030a60 bound to <yate server ip>:39018 +RTCP [0x7f98fc0419a0]
<yrtp:ALL> YRTPSource::YRTPSource(0x7f98fc0419a0) [0x7f98fc005940]
<yrtp:ALL> YRTPConsumer::YRTPConsumer(0x7f98fc0419a0) [0x7f98fc007e80]
<yrtp:ALL> YRTPWrapper::setupSRTP(true) [0x7f98fc0419a0]
<yrtp:ALL> RTP/SAVP message received
<yrtp:ALL> Wrapper 0x7f98fc0419a0 found by CallEndpoint 0x7f98fc02e760
<yrtp:ALL> YRTPWrapper::startRTP("<phone ip 2>",53388) [0x7f98fc0419a0]
<yrtp:INFO> RTP starting format 'alaw' payload 8 [0x7f98fc0419a0]
<INFO> RTPSecure::init() encrypt=true authlen=4 [0x7f98fc005d60]
<yrtp:ALL> YRTPWrapper::startSRTP('AES_CM_128_HMAC_SHA1_32','inline:fXFTe5zhvpd9j85nnjDTXVRwCtyi/eHI82gB/NVp',(nil)) [0x7f98fc0419a0]
<INFO> RTPSecure::setup('AES_CM_128_HMAC_SHA1_32','inline:fXFTe5zhvpd9j85nnjDTXVRwCtyi/eHI82gB/NVp',(nil)) [0x7f98e805fb50]
<INFO> RTPSecure::init() encrypt=true authlen=4 [0x7f98e805fb50]
<yrtp:NOTE> Started SRTP suite 'AES_CM_128_HMAC_SHA1_32' [0x7f98fc0419a0]
<yrtp:ALL> YRTPWrapper::setupSRTP(true) [0x7f98fc0419a0]
<NOTE> Choosing started 'audio' format 'alaw' [0x7f98fc030080]
<yrtp:ALL> RTP/SAVP message received
<yrtp:ALL> YRTPWrapper::YRTPWrapper('<yate server ip>',0x7f98e8060140,'audio',bidir,0x7f98fc002400,false) [0x7f98fc02f900]
<yrtp:ALL> YRTPWrapper::setupRTP("<yate server ip>",true,true) [0x7f98fc02f900]
<yrtp:INFO> Session 'yrtp/1232061648' 0x7f98fc04bd60 bound to <yate server ip>:39708 +RTCP [0x7f98fc02f900]
<yrtp:ALL> YRTPSource::YRTPSource(0x7f98fc02f900) [0x7f98fc006f50]
<INFO> DataTranslator::attachChain [0x7f98fc006f50] '(null)' -> [0x7f98fc007e80] 'alaw' not possible
<yrtp:ALL> YRTPConsumer::YRTPConsumer(0x7f98fc02f900) [0x7f98fc0070e0]
<INFO> DataTranslator::attachChain [0x7f98fc005940] 'alaw' -> [0x7f98fc0070e0] '(null)' not possible
<yrtp:ALL> YRTPWrapper::startRTP("<phone ip 1>",50726) [0x7f98fc02f900]
<yrtp:INFO> RTP starting format 'alaw' payload 8 [0x7f98fc02f900]
>>> DataTranslator::detachChain(0x7f98fc006f50,0x7f98fc007e80)
<<< DataTranslator::detachChain
<ALL> DataTranslator::attachChain [0x7f98fc006f50] 'alaw' -> [0x7f98fc007e80] 'alaw' succeeded
>>> DataTranslator::detachChain(0x7f98fc005940,0x7f98fc0070e0)
<<< DataTranslator::detachChain
<ALL> DataTranslator::attachChain [0x7f98fc005940] 'alaw' -> [0x7f98fc0070e0] 'alaw' succeeded
<yrtp:ALL> YRTPWrapper::startSRTP('AES_CM_128_HMAC_SHA1_32','inline:69Lro0DKDtf+bZyA3z8St0wYyCc7yzdNWKDDwSJC',(nil)) [0x7f98fc02f900]
<INFO> RTPSecure::setup('AES_CM_128_HMAC_SHA1_32','inline:69Lro0DKDtf+bZyA3z8St0wYyCc7yzdNWKDDwSJC',(nil)) [0x7f98fc005550]
<INFO> RTPSecure::init() encrypt=true authlen=4 [0x7f98fc005550]
<yrtp:NOTE> Started SRTP suite 'AES_CM_128_HMAC_SHA1_32' [0x7f98fc02f900]
<yrtp:ALL> YRTPWrapper::setupSRTP(true) [0x7f98fc02f900]
<INFO> RTPSecure::init() encrypt=true authlen=4 [0x7f98fc03c850]
<NOTE> Choosing started 'audio' format 'alaw' [0x7f98e805ef10]
<yrtp:ALL> RTP/SAVP message received
<yrtp:ALL> Wrapper 0x7f98fc02f900 found by CallEndpoint 0x7f98e8060140
<yrtp:ALL> YRTPWrapper::startRTP("<phone ip 1>",50726) [0x7f98fc02f900]


[abma]

i've tried to simplify the testcase:

in regexroute with this:

Code: [Select]

^99$=wave/record//tmp/echoback-${caller}.slin;

the file gets created, but its empty when i hang up the call. an idea what could be wrong / why yate doesn't receive (or understand?) the rtp from the phone?

Code: [Select]

20160502175741.345343 <INFO> Could not classify call from '<caller>', wasted 12 usec
20160502175741.346042 <INFO> Routing call to '99' in context 'default' via 'wave/record//tmp/echoback-<caller>.slin' in 300 usec
20160502175741.346534 <wave:INFO> Record to wave file '/tmp/echoback-<caller>.slin'
20160502175741.347064 <wave/47:ALL> WaveChan::WaveChan(record) [0x7f7ef00511e0]
20160502175741.347355 <wave:ALL> WaveConsumer::WaveConsumer("/tmp/echoback-<caller>.slin",0x7f7ef00511e0,0,"(null)",false,0x7f7ef0035eb0) [0x7f7ef00077c0]
20160502175741.348078 <yrtp:ALL> RTP/SAVP message received
20160502175741.348227 <yrtp:INFO> Guessed local IP '<pbx ip>' for remote '<phone ip>'
20160502175741.348277 <yrtp:ALL> YRTPWrapper::YRTPWrapper('<pbx ip>',0x7f7f0c0764f0,'audio',bidir,0x7f7ef0055c20,false) [0x7f7ef0018510]
20160502175741.348298 <yrtp:ALL> YRTPWrapper::setupRTP("<pbx ip>",true,true) [0x7f7ef0018510]
20160502175741.348364 <yrtp:INFO> Session 'yrtp/1841441955' 0x7f7ef00516b0 bound to <pbx ip>:28468 +RTCP [0x7f7ef0018510]
20160502175741.348409 <yrtp:ALL> YRTPSource::YRTPSource(0x7f7ef0018510) [0x7f7ef0050c70]
20160502175741.348429 <INFO> DataTranslator::attachChain [0x7f7ef0050c70] '(null)' -> [0x7f7ef00077c0] 'slin' not possible
20160502175741.348446 <yrtp:ALL> YRTPConsumer::YRTPConsumer(0x7f7ef0018510) [0x7f7ef0024b30]
20160502175741.348468 <yrtp:ALL> YRTPWrapper::startRTP("<phone ip>",52072) [0x7f7ef0018510]
20160502175741.348491 <yrtp:INFO> RTP starting format 'alaw' payload 8 [0x7f7ef0018510]
20160502175741.348513 >>> DataTranslator::detachChain(0x7f7ef0050c70,0x7f7ef00077c0)
20160502175741.348535 <<< DataTranslator::detachChain
20160502175741.348553 <ALL> Created DataTranslator 0x7f7ef00148e0 for 'alaw' -> 'slin' by factory 0x7f7f3985b980 (len=1)
20160502175741.348864 <ALL> DataTranslator::attachChain [0x7f7ef0050c70] 'alaw' -> [0x7f7ef00077c0] 'slin' succeeded
20160502175741.349322 <yrtp:ALL> YRTPWrapper::startSRTP('AES_CM_128_HMAC_SHA1_32','inline:XaoDgX8KDCLdwMJ3zS2fT+/EDUeQ42G0VXSnZt7D',(nil)) [0x7f7ef0018510]
20160502175741.349947 <INFO> RTPSecure::setup('AES_CM_128_HMAC_SHA1_32','inline:XaoDgX8KDCLdwMJ3zS2fT+/EDUeQ42G0VXSnZt7D',(nil)) [0x7f7ef0054900]
20160502175741.351158 <INFO> RTPSecure::init() encrypt=true authlen=4 [0x7f7ef0054900]
20160502175741.351428 <yrtp:NOTE> Started SRTP suite 'AES_CM_128_HMAC_SHA1_32' [0x7f7ef0018510]
20160502175741.351609 <yrtp:ALL> YRTPWrapper::setupSRTP(true) [0x7f7ef0018510]
20160502175741.351684 <INFO> RTPSecure::init() encrypt=true authlen=4 [0x7f7ef0024920]
20160502175741.351757 <NOTE> Choosing started 'audio' format 'alaw' [0x7f7f0c02ac00]
20160502175741.416401 <yrtp:ALL> RTP/SAVP message received
20160502175741.416563 <yrtp:ALL> Wrapper 0x7f7ef0018510 found by CallEndpoint 0x7f7f0c0764f0
20160502175741.416588 <yrtp:ALL> YRTPWrapper::startRTP("<phone ip>",52072) [0x7f7ef0018510]


[abma]

more details:

its a snom715: when i disable SRTP it works, with SRTP enabled nothing gets recorded.

[abma]

and more details

in "TelEngine::RTPReceiver::rtpData(const void*, int)" rtpCheckIntegrity() fails.

Code: [Select]

2016-05-04_16:00:34.969001 <WARN> SRTP HMAC recv: e4dfae70 calc: 0a1071eb seq: 66331 [0x7f47c801d5a0]

an idea what could cause this / how to solve this?

[abma]

(very likely) a bug in yate:

http://forum.yate.ro/index.php?topic=1273.0