sipML5 and Freeswitch

[PeterKrause]

With the live demo, I can register to my personal FreeSwitch server
with no problems. When I try to make a call, FS complains:

a=crypto in RTP/AVP, refer to rfc3711

and hangs up on me. In the INVITE, sipML5 is sending:

a=crypto:0 AES_CM_128_HMAC_SHA1_80 inline:RIxXHNE/
5GtDfkXnAV4CLXBQGxPzlzBg3Xyg3Lfs

According to rfc3711, crypto shouldn't be on an a= line, I believe m=
is possible.

Is there a way in the live demo to disable the crypto line? Or is
this coming from the library itself?

Thanks
Peter

[Mamadou]

No "a=crypto" is correct. See rfc 4568.
SRTP encryption is mandatory in WebRTC.

[Meftah Tayeb]

Peter,
who you're calling on the freeswitch, on the other side ?
would be good if you ofer a full sip traces

__________ Information from ESET NOD32 Antivirus, version of virus signature
database 6830 (20120126) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




__________ Information from ESET NOD32 Antivirus, version of virus signature database 6830 (20120126) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

[pe...@uringme.com]

I'm calling a local extension on my FreeSwitch server, 7779, which currently just plays a voice prompt.


I believe I have mis-read some earlier info from FreeSwitch, which is why I thought a=crypto is not allowed. I see that it is, but I'm still sort of stuck because I can't seem to get a call to work.

Below are two links that I was looking through for crypto compliance. They're a few years old, but I was trying to find some authoritative statement from Brian West about FreeSwitch and crypto compliance.

Following the links is my Invite from the live demo, and FreeSwitch's response.

Thanks for your responses.

http://lists.freeswitch.org/pipermail/freeswitch-dev/2008-March/000864.html
http://lists.freeswitch.org/pipermail/freeswitch-users/2008-April/030773.html



INVITE sip:77...@65.110.58.60 SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:4060;branch=z9hG4bK-524287-1---a56eca302b8d6412;rport
Via: SIP/2.0/TCP 68.205.137.246:62676;branch=z9hG4bK1P36uFnLdbzHiEzWdVAd0G6nk6kdOdEt;rport=62676;received=68.205.137.246
Max-Forwards: 69
Contact: "Peter Krause"<sip:77...@87.106.69.240:4060;transport=udp;ws-src-ip=68.205.137.246;ws-src-port=62676>;+sip.ice
To: <sip:77...@65.110.58.60>
From: <sip:7777@uringme>;tag=gz6bNBeAkHM7eAR2eoJG
Call-ID: 434d70e1-5cc1-d096-442c-499c8cd600ad
CSeq: 6437 INVITE
Content-Type: application/sdp
Organization: Doubango Telecom
User-Agent: IM-client/OMA1.0 sipML5/v0.0.0000.0
Content-Length: 3483

v=0
o=- 275367574 1 IN IP4 127.0.0.1
s=webrtc (chrome 20.0.1127.0) - Doubango Telecom (sipML5 r000)
t=0 0
m=audio 54450 RTP/SAVPF 103 104 0 8 106 105 13 126
c=IN IP4 68.205.137.246
a=rtcp:54451 IN IP4 68.205.137.246
a=candidate:0 1 udp 2130706432 192.168.1.8 50256 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0
a=candidate:0 2 udp 2130706432 192.168.1.8 50257 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0
a=candidate:0 1 udp 1912602624 68.205.137.246 54450 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0
a=candidate:0 2 udp 1912602624 68.205.137.246 54451 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0
a=candidate:0 1 tcp 1694498816 192.168.1.8 62677 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0
a=candidate:0 2 tcp 1694498816 192.168.1.8 62678 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0
a=mid:audio
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:33ayFPGAoi+nni8kbYZfBO7tH3qI1SaH7ilFU7GT
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Hd6rAXtyms1rJgC4OTJk/K4AGpnwLUn9Aqdx1sbd
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=ssrc:2075114348 cname:vcjaT0z7bAvF1TL3
a=ssrc:2075114348 mslabel:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na
a=ssrc:2075114348 label:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na0
a=sendrecv
m=video 54452 RTP/SAVPF 100 101 102
c=IN IP4 68.205.137.246
a=rtcp:54453 IN IP4 68.205.137.246
a=candidate:0 1 udp 2130706432 192.168.1.8 50258 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0
a=candidate:0 2 udp 2130706432 192.168.1.8 50259 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0
a=candidate:0 1 udp 1912602624 68.205.137.246 54452 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0
a=candidate:0 2 udp 1912602624 68.205.137.246 54453 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0
a=candidate:0 1 tcp 1694498816 192.168.1.8 62679 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0
a=candidate:0 2 tcp 1694498816 192.168.1.8 62680 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0
a=mid:video
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_80 inline:RIxXHNE/5GtDfkXnAV4CLXBQGxPzlzBg3Xyg3Lfs
a=rtpmap:100 VP8/90000
a=rtpmap:101 red/90000
a=rtpmap:102 ulpfec/90000
a=ssrc:2075114348 cname:vcjaT0z7bAvF1TL3
a=ssrc:2075114348 mslabel:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na
a=ssrc:2075114348 label:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na0
a=sendrecv
------------------------------------------------------------------------
send 448 bytes to udp/[87.106.69.240]:4060 at 17:04:47.774032:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 87.106.69.240:4060;branch=z9hG4bK-524287-1---a56eca302b8d6412;rport=4060
Via: SIP/2.0/TCP 68.205.137.246:62676;branch=z9hG4bK1P36uFnLdbzHiEzWdVAd0G6nk6kdOdEt;rport=62676;received=68.205.137.246
From: <sip:7777@uringme>;tag=gz6bNBeAkHM7eAR2eoJG
To: <sip:77...@65.110.58.60>
Call-ID: 434d70e1-5cc1-d096-442c-499c8cd600ad
CSeq: 6437 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-
Content-Length: 0

------------------------------------------------------------------------
2012-05-21 13:04:47.754843 [NOTICE] switch_channel.c:816 New Channel sofia/external/7777@uringme [120f19be-a367-11e1-b22e-57011ca30f08]
2012-05-21 13:04:47.754843 [DEBUG] sofia.c:4770 Channel sofia/external/7777@uringme entering state [received][100]
2012-05-21 13:04:47.754843 [DEBUG] sofia.c:4781 Remote SDP:
v=0
o=- 275367574 1 IN IP4 127.0.0.1
s=webrtc (chrome 20.0.1127.0) - Doubango Telecom (sipML5 r000)
t=0 0
m=audio 54450 RTP/SAVPF 103 104 0 8 106 105 13 126
c=IN IP4 68.205.137.246
a=rtcp:54451 IN IP4 68.205.137.246
a=candidate:0 1 udp 2130706432 192.168.1.8 50256 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0
a=candidate:0 2 udp 2130706432 192.168.1.8 50257 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0
a=candidate:0 1 udp 1912602624 68.205.137.246 54450 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0
a=candidate:0 2 udp 1912602624 68.205.137.246 54451 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0
a=candidate:0 1 tcp 1694498816 192.168.1.8 62677 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0
a=candidate:0 2 tcp 1694498816 192.168.1.8 62678 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0
a=mid:audio
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:33ayFPGAoi+nni8kbYZfBO7tH3qI1SaH7ilFU7GT
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Hd6rAXtyms1rJgC4OTJk/K4AGpnwLUn9Aqdx1sbd
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=ssrc:2075114348 cname:vcjaT0z7bAvF1TL3
a=ssrc:2075114348 mslabel:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na
a=ssrc:2075114348 label:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na0
m=video 54452 RTP/SAVPF 100 101 102
c=IN IP4 68.205.137.246
a=rtcp:54453 IN IP4 68.205.137.246
a=candidate:0 1 udp 2130706432 192.168.1.8 50258 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0
a=candidate:0 2 udp 2130706432 192.168.1.8 50259 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0
a=candidate:0 1 udp 1912602624 68.205.137.246 54452 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0
a=candidate:0 2 udp 1912602624 68.205.137.246 54453 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0
a=candidate:0 1 tcp 1694498816 192.168.1.8 62679 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0
a=candidate:0 2 tcp 1694498816 192.168.1.8 62680 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0
a=mid:video
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_80 inline:RIxXHNE/5GtDfkXnAV4CLXBQGxPzlzBg3Xyg3Lfs
a=rtpmap:100 VP8/90000
a=rtpmap:101 red/90000
a=rtpmap:102 ulpfec/90000
a=ssrc:2075114348 cname:vcjaT0z7bAvF1TL3
a=ssrc:2075114348 mslabel:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na
a=ssrc:2075114348 label:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na0

2012-05-21 13:04:47.754843 [ERR] sofia_glue.c:4478 a=crypto in RTP/AVP, refer to rfc3711
2012-05-21 13:04:47.754843 [DEBUG] switch_channel.c:2592 (sofia/external/7777@uringme) Callstate Change DOWN -> HANGUP
2012-05-21 13:04:47.754843 [NOTICE] sofia.c:4998 Hangup sofia/external/7777@uringme [CS_NEW] [INCOMPATIBLE_DESTINATION]




--- On Sun, 5/20/12, Meftah Tayeb <tayeb....@gmail.com> wrote:

[Mamadou]

Your logs say that the server don't support RTP/SAVPF profile and
except RTP/SAVP. Your server should ignore the profile as the media
will be p2p (one of ICE candidates will be used).
You can also hack the SDP like this:

var o_hdr_M = this.o_sdp_lo.get_header_m_by_name("audio");
if (o_hdr_M)o_hdr_m.s_proto = "RTP/SAVP";
o_hdr_M = this.o_sdp_lo.get_header_m_by_name("video");
if (o_hdr_M)o_hdr_m.s_proto = "RTP/SAVP";

Source code to be added here: http://code.google.com/p/sipml5/source/browse/trunk/src/tinyMEDIA/src...
Note that this will disable RTCP-FB on the remote which means that you
will have bad or no video at all.

On May 21, 7:53 pm, pe...@uringme.com wrote:
> I'm calling a local extension on my FreeSwitch server, 7779, which currently just plays a voice prompt.
>
> I believe I have mis-read some earlier info from FreeSwitch, which is why I thought a=crypto is not allowed.  I see that it is, but I'm still sort of stuck because I can't seem to get a call to work.
>
> Below are two links that I was looking through for crypto compliance.  They're a few years old, but I was trying to find some authoritative statement from Brian West about FreeSwitch and crypto compliance.
>
> Following the links is my Invite from the live demo, and FreeSwitch's response.
>
> Thanks for your responses.
>

> http://lists.freeswitch.org/pipermail/freeswitch-dev/2008-March/00086...http://lists.freeswitch.org/pipermail/freeswitch-users/2008-April/030...
>
>    INVITE sip:7...@65.110.58.60 SIP/2.0

>    Via: SIP/2.0/UDP 87.106.69.240:4060;branch=z9hG4bK-524287-1---a56eca302b8d6412;rport
>    Via: SIP/2.0/TCP 68.205.137.246:62676;branch=z9hG4bK1P36uFnLdbzHiEzWdVAd0G6nk6kdOdEt;rport=62676;received=68.205.137.246
>    Max-Forwards: 69

>    Contact: "Peter Krause"<sip:7...@87.106.69.240:4060;transport=udp;ws-src-ip=68.205.137.246;ws-src-port=62676>;+sip.ice
>    To: <sip:7...@65.110.58.60>

>    To: <sip:7...@65.110.58.60>

> 2012-05-21 13:04:47.754843 [NOTICE] sofia.c:4998 Hangup sofia/external/7777@uringme [CS_NEW] [INCOMPATIBLE_DESTINATION]...
>
> read more »

[pe...@uringme.com]

I made the SAVP change. If an endpoint does not support video, is sipML5 supposed to handle that automatically? I'm just dialing an extension that plays an audio message (no video), but I get a DOM exception.

This is the Java trace for the call after I made the SAVPF to SAVP change:

SIP stack start: proxy='sipml5.org:4062', realm='<sip:65.110.58.60>', impi='7777', impu='<sip:7777@uringme>' tsip_stack.js:353
Connecting to 'ws://sipml5.org:4062' tsip_transport.js:326
Stack starting call.htm:424
__tsip_transport_ws_onopen tsip_transport.js:363
Stack started call.htm:424
State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister tsk_fsm.js:99
Connecting... call.htm:471
REGISTER request successfully sent call.htm:471
__tsip_transport_ws_onmessage tsip_transport.js:375
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 tsk_fsm.js:99
REGISTER request successfully sent call.htm:471
__tsip_transport_ws_onmessage tsip_transport.js:375
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx tsk_fsm.js:99
Connected call.htm:471
__tsip_transport_ws_onmessage tsip_transport.js:375
State machine: c0000_Started_2_Outgoing_X_oINVITE tsk_fsm.js:99
__on_state_change tmedia_session_jsep.js:103
Media Added call.htm:559
Call in progress... call.htm:471
13__on_ice_candidate: 1792 tmedia_session_jsep.js:85
Start SAVP hack tmedia_session_jsep.js:188
o_hdr_M=RTP/SAVPF tmedia_session_jsep.js:190
__tsip_transport_ws_onmessage tsip_transport.js:375
State machine: x0000_Any_2_Any_X_i1xx tsk_fsm.js:99
Trying call.htm:559
__tsip_transport_ws_onmessage tsip_transport.js:375
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE tsk_fsm.js:99
SDP_RO=v=0
o=FreeSWITCH 1337758058 1337758059 IN IP4 65.110.58.60
s=FreeSWITCH
c=IN IP4 65.110.58.60
t=0 0
m=audio 27680 RTP/SAVP 0 126 13
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=rtpmap:13 CN/8000
a=ptime:20
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:rMS9onCocnQO+wFcjmakmG6e1GGT1pafl8vqxezL
m=video 0 RTP/SAVP 19
tmedia_session_jsep.js:234
b_is_offer=false tmedia_session_jsep.js:235
DOMException
code: 12
message: "SYNTAX_ERR: DOM Exception 12"
name: "SYNTAX_ERR"
stack: "Error: An invalid or illegal string was specified.↵ at tmedia_session.__set_ro (file:///C:/uringme/starpound/sipml5/trunk/src/tinyMEDIA/src/tmedia_session_jsep.js:238:23)↵ at tmedia_session.set_ro (file:///C:/uringme/starpound/sipml5/trunk/src/tinyMEDIA/src/tmedia_session.js:565:17)↵ at tmedia_session_mgr.set_ro (file:///C:/uringme/starpound/sipml5/trunk/src/tinyMEDIA/src/tmedia_session.js:387:29)↵ at tsip_dialog.process_ro (file:///C:/uringme/starpound/sipml5/trunk/src/tinySIP/src/dialogs/tsip_dialog_invite.js:745:39)↵ at tsk_fsm_entry.c0000_Outgoing_2_Connected_X_i2xxINVITE [as fn_execute] (file:///C:/uringme/starpound/sipml5/trunk/src/tinySIP/src/dialogs/tsip_dialog_invite__client.js:131:27)↵ at tsk_fsm.act (file:///C:/uringme/starpound/sipml5/trunk/src/tinySAK/src/tsk_fsm.js:106:47)↵ at tsip_dialog.fsm_act (file:///C:/uringme/starpound/sipml5/trunk/src/tinySIP/src/dialogs/tsip_dialog.js:739:23)↵ at
tsip_dialog.__tsip_dialog_invite_event_callback [as fn_callback] (file:///C:/uringme/starpound/sipml5/trunk/src/tinySIP/src/dialogs/tsip_dialog_invite.js:894:26)↵ at tsip_dialog.callback (file:///C:/uringme/starpound/sipml5/trunk/src/tinySIP/src/dialogs/tsip_dialog.js:724:21)↵ at tsk_fsm_entry.__tsip_transac_ict_Proceeding_2_Accepted_X_2xx [as fn_execute] (file:///C:/uringme/starpound/sipml5/trunk/src/tinySIP/src/transactions/tsip_transac_ict.js:469:32)"
__proto__: DOMException
tmedia_session_jsep.js:245
tmedia_session_jsep.__set_ro tmedia_session_jsep.js:245
tmedia_session.set_ro tmedia_session.js:565
tmedia_session_mgr.set_ro tmedia_session.js:387
tsip_dialog_invite.process_ro tsip_dialog_invite.js:745
c0000_Outgoing_2_Connected_X_i2xxINVITE tsip_dialog_invite__client.js:131
tsk_fsm.act tsk_fsm.js:106
tsip_dialog.fsm_act tsip_dialog.js:739
__tsip_dialog_invite_event_callback tsip_dialog_invite.js:894
tsip_dialog.callback tsip_dialog.js:724
__tsip_transac_ict_Proceeding_2_Accepted_X_2xx tsip_transac_ict.js:469
tsk_fsm.act tsk_fsm.js:106
tsip_transac.fsm_act tsip_transac.js:138
__tsip_transac_ict_event_callback tsip_transac_ict.js:570
tsip_transac.callback tsip_transac.js:100
tsip_transac_layer.handle_incoming_message tsip_transac_layer.js:277
tsip_transport_layer.handle_incoming_message tsip_transport_layer.js:238
__tsip_transport_ws_onmessage tsip_transport.js:389
OK call.htm:559
In Call call.htm:471

--- On Mon, 5/21/12, Mamadou <diopm...@doubango.org> wrote:

> From: Mamadou <diopm...@doubango.org>
> Subject: Re: sipML5 and Freeswitch

[Mamadou]

This a bug in chrome (http://groups.google.com/group/discuss-webrtc/
browse_thread/thread/6212397d4a71d637) caused by the fact that the
video port is equal to zero.
Like the profile, try to patch the SDP.

> --- On Mon, 5/21/12, Mamadou <diopmama...@doubango.org> wrote:

[pe...@uringme.com]

Okay, changed so only audio, no video.

I get a connection, and FreeSwitch is sending RTP data, but I don't hear any audio. I see it going to my public IP, but no audio. I tried the same extension with another SIP client (X-lite) and that worked, so I know there are no firewall/routing issues.

I noticed in the INVITE SDP that the origin line is 127.0.0.1 (it was my public IP in the X-Lite INVITE). The c= line is fine, though. Is there an API to change the IP of the o= line in the SDP?

INVITE and first part of SDP follows:

INVITE sip:77...@65.110.58.60 SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:7060;branch=z9hG4bK-524287-1---28761c25c112c562;rport
Via: SIP/2.0/TCP 68.205.137.246:49839;branch=z9hG4bKD2WhvHpVoTife74BXNJ7voPwvcQRyUZk;rport=49839;received=68.205.137.246
Max-Forwards: 69
Contact: "Peter Krause"<sip:77...@87.106.69.240:7060;transport=udp;ws-src-ip=68.205.137.246;ws-src-port=49839>;+sip.ice
To: <sip:77...@65.110.58.60>
From: <sip:7777@uringme>;tag=E9uN7TzQ1obmdOUuMFLD
Call-ID: c26cd184-eb23-e565-bfd3-0bfafe19742e
CSeq: 13372 INVITE

Content-Type: application/sdp
Organization: Doubango Telecom
User-Agent: IM-client/OMA1.0 sipML5/v0.0.0000.0

Content-Length: 1359

v=0
o=- 2660658294 1 IN IP4 127.0.0.1

s=webrtc (chrome 20.0.1127.0) - Doubango Telecom (sipML5 r000)

a=group:BUNDLE audio video
t=0 0
m=audio 50493 RTP/SAVP 103 104 0 8 106 105 13 126
c=IN IP4 68.205.137.246
....snip....

--- On Wed, 5/23/12, Mamadou <diopm...@doubango.org> wrote:

[Mamadou]

Chrome (and any browser supporting WebRTC) requires ICE and SRTP.
Your server must retransmit both STUN and SRTP packets.

On May 23, 10:07 pm, pe...@uringme.com wrote:
> Okay, changed so only audio, no video.
>
> I get a connection, and FreeSwitch is sending RTP data, but I don't hear any audio.  I see it going to my public IP, but no audio.  I tried the same extension with another SIP client (X-lite) and that worked, so I know there are no firewall/routing issues.
>
> I noticed in the INVITE SDP that the origin line is 127.0.0.1 (it was my public IP in the X-Lite INVITE).  The c= line is fine, though.  Is there an API to change the IP of the o= line in the SDP?
>
> INVITE and first part of SDP follows:
>

>    INVITE sip:7...@65.110.58.60 SIP/2.0

>    Via: SIP/2.0/UDP 87.106.69.240:7060;branch=z9hG4bK-524287-1---28761c25c112c562;rport
>    Via: SIP/2.0/TCP 68.205.137.246:49839;branch=z9hG4bKD2WhvHpVoTife74BXNJ7voPwvcQRyUZk;rport=49839;received=68.205.137.246
>    Max-Forwards: 69

>    Contact: "Peter Krause"<sip:7...@87.106.69.240:7060;transport=udp;ws-src-ip=68.205.137.246;ws-src-port=49839>;+sip.ice
>    To: <sip:7...@65.110.58.60>

>    From: <sip:7777@uringme>;tag=E9uN7TzQ1obmdOUuMFLD
>    Call-ID: c26cd184-eb23-e565-bfd3-0bfafe19742e
>    CSeq: 13372 INVITE
>    Content-Type: application/sdp
>    Organization: Doubango Telecom
>    User-Agent: IM-client/OMA1.0 sipML5/v0.0.0000.0
>    Content-Length: 1359
>
>    v=0
>    o=- 2660658294 1 IN IP4 127.0.0.1
>    s=webrtc (chrome 20.0.1127.0) - Doubango Telecom (sipML5 r000)
>    a=group:BUNDLE audio video
>    t=0 0
>    m=audio 50493 RTP/SAVP 103 104 0 8 106 105 13 126
>    c=IN IP4 68.205.137.246
> ....snip....
>

[pe...@uringme.com]

I'm using ICE and SRTP, not STUN. My server is the endpoint -- it answers and plays a prompt (like an IVR), it doesn't route. I was thinking the audio couldn't get routed back to me because the o= line is 127.0.0.1 even though ICE shows the correct public IP of my home router. tcpdump shows packets going from freeswitch to my router, but WireShark shows nothing coming to my PC from inside the router, so something is getting lost between the router and my PC, which sounds like an SDP issue.

When I use another SIP client (X-Lite, not Chrome) and dial the same extension on the same server, I get audio no problem.

Maybe the o= line isn't the problem, but it's a major difference in the two INVITEs.

--- On Wed, 5/23/12, Mamadou <diopm...@doubango.org> wrote:

[pe...@uringme.com]

I might have figured this out. I'm running the web page locally through file:// instead of http:// on my windows box. That's probably why it's 127.0.0.1. It's using lo instead of a real NIC.

--- On Wed, 5/23/12, pe...@uringme.com <pe...@uringme.com> wrote:

[Mamadou]

You're trying to connect to a private server (10.0.0.4) from the
publicly hosted gateway (sipml5.org:4062) and this is why it doesn't
work.
Three solutions:
1- Host a WebSocket gateway on the same private network
2- Put your server on the public domain
3- Use a SIP server supporting WebSocket transport

On Jun 5, 11:25 am, Bradly Swart <brad8...@gmail.com> wrote:
> Hi Peter,
>
> Seeing as though you have done this before I was hoping that you could help
> me out here.
> I can't seem to get the SIPML5 Demo to register to my Freeswitch box, did
> you have to do anything special ?
>
> I can register and call between two softphones, so it looks like the
> standard Freeswitch setup is working.
> The registration from the sipml5 demo doesn't seem to hit the FS server.
> Javascript output is;
>
> SIP stack start: proxy='sipml5.org:4062', realm='<sip:10.0.0.4>',
> impi='1001', impu='<sip:1...@10.0.0.4>' tsk_utils.js:97<http://10.0.0.4/sipml5/src/tinySAK/src/tsk_utils.js?svn=5>
> Connecting to 'ws://sipml5.org:4062' tsk_utils.js:97<http://10.0.0.4/sipml5/src/tinySAK/src/tsk_utils.js?svn=5>
> Stack starting tsk_utils.js:97<http://10.0.0.4/sipml5/src/tinySAK/src/tsk_utils.js?svn=5>
> __tsip_transport_ws_onopen tsk_utils.js:97<http://10.0.0.4/sipml5/src/tinySAK/src/tsk_utils.js?svn=5>
> Stack started tsk_utils.js:97<http://10.0.0.4/sipml5/src/tinySAK/src/tsk_utils.js?svn=5>
> State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister
> tsk_utils.js:97 <http://10.0.0.4/sipml5/src/tinySAK/src/tsk_utils.js?svn=5>
> Connecting... tsk_utils.js:97<http://10.0.0.4/sipml5/src/tinySAK/src/tsk_utils.js?svn=5>
> REGISTER request successfully sent tsk_utils.js:97<http://10.0.0.4/sipml5/src/tinySAK/src/tsk_utils.js?svn=5>
> __tsip_transport_ws_onmessage tsk_utils.js:97<http://10.0.0.4/sipml5/src/tinySAK/src/tsk_utils.js?svn=5>
> State machine: tsip_dialog_register_InProgress_2_InProgress_X_1xx
> tsk_utils.js:97 <http://10.0.0.4/sipml5/src/tinySAK/src/tsk_utils.js?svn=5>
> <http://10.0.0.4/sipml5/src/tinySAK/src/tsk_utils.js?svn=5>
>
> Any help would be much appreciated!

>
>
>
>
>
>
>
> On Sunday, 20 May 2012 17:47:58 UTC+2, Meftah tayeb wrote:
>
> > Peter,
> > who you're calling on the freeswitch, on the other side ?
> > would be good if you ofer a full sip traces
>

> > ----- Original Message -----
> > From: "Mamadou" <diopmama...@doubango.org>
> > To: "Doubango" <doub...@googlegroups.com>
> > Sent: Monday, May 21, 2012 8:32 PM

> > Subject: Re: sipML5 and Freeswitch
>

> > No "a=crypto" is correct. See rfc 4568.
> > SRTP encryption is mandatory in WebRTC.
>
> > On May 21, 7:29 pm, PeterKrause <pe...@uringme.com> wrote:
> > > With the live demo, I can register to my personal FreeSwitch server
> > > with no problems. When I try to make a call, FS complains:
>

> > > a=crypto in RTP/AVP, refer to rfc3711
>

[Bradly Swart]

Great thanks for pointing that out, I tried playing around with that without realizing what it was actually doing.

For interest sake what are you using as your socket server ? I'm going to look at implementing a Node.js one alongside my apache.

--

Bradly Swart

Mobile: +27 829-710-821

Skype: bradly.swart

Twitter: @brad8711

[Mamadou]

I'm using webrtc2sip (http://code.google.com/p/webrtc2sip/) but any
SIP server with support for WebSocket should work

[Johnson Huang]

Hi all,

For this issue ,we need to setup the TLS/SRTP in freeswitch server then can solve "a=crypto in RTP/AVP, refer to rfc3711" error ?

Johnson Huang

PeterKrause於 2012年5月22日星期二UTC+8上午1時29分47秒寫道：

[Muhammad Shahzad]

Yes, please lift this restriction, since it is becoming a big hurdle in integrating FS with WebRTC clients.

Thank you.

On Fri, Jul 20, 2012 at 6:49 PM, <anthony....@gmail.com> wrote:

The problem is not with the crypto its with the a=crypto being inside the AVP vs SAVP (denoting secure)

see http://jira.freeswitch.org/browse/FS-636

I am willing to lift this restriction or at least make it configurable since the alternative is you must send a double sized sdp with AVP for non secure stuff and SAVP for the secure.

--
 
 

--
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari...@hotmail.com
Email: shahe...@googlemail.com

[Muhammad Shahzad]

Slightly unrelated to this mailing list but i think VP8 codec support is pending in FS for over 3 months. A patch was submitted by Seven Du on April 09, 2012 but it is i think still pending merge into FS trunk,

http://jira.freeswitch.org/browse/FS-4092

Anthony, can you please get it merged, so we have better support for WebRTC and doubango clients in FS. It would be a big plus for FS since there is no other media server that i know which has VP8 support.

Thank you.

[Juan Vasquez]

All

what exact files need to be hacked to accomplish the hack to work with freeswitch?

from the previous post i see the suggestion, but not sure exactly in what file.

Regards

-Juan

You can also hack the SDP like this: 

var o_hdr_M = this.o_sdp_lo.get_header_m_by_name("audio"); 
if (o_hdr_M)o_hdr_m.s_proto = "RTP/SAVP"; 
o_hdr_M = this.o_sdp_lo.get_header_m_by_name("video"); 
if (o_hdr_M)o_hdr_m.s_proto = "RTP/SAVP"; 

Source code to be added here: http://code.google.com/p/sipml5/source/browse/trunk/src/tinyMEDIA/src... 

[Mamadou DIOP]

 http://code.google.com/p/webrtc2sip/#Testing_the_gateway

On Nov 30, 2012, at 9:58 PM, Ros P <rosph...@gmail.com> wrote:

Ive been able to successfully register a device using webrtc2sip gw, however i cant get any sound. The webserver, gw, and freeswitch are all on the same machine. Any ideas would be greatly appreciated?

State machine: x0000_Any_2_Any_X_i401_407_INVITE /src/tinySAK/src/tsk_utils.js?svn=13:59

SEND: INVITE sip:9999...@107.23.59.74 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcK4Ws75sM3nZIqUmq3nK5XOHdPtRcFiN;rport From: <sip:21303...@107.23.59.74>;tag=TIQtKygCbCAPoVtU7DpO To: <sip:9999...@107.23.59.74> Contact: "sp"<sip:21303...@df7jal23ls0d.invalid;transport=ws>;+sip.ice Call-ID: 7de16dc9-41b9-b1c6-b085-1b891489813e CSeq: 16406 INVITE Content-Type: application/sdp Content-Length: 1294 Max-Forwards: 70 Proxy-Authorization: Digest username="2130334003",realm="107.23.59.74",nonce="550d599a-3b30-11e2-a729-8b18d47feaee",uri="sip:9999...@107.23.59.74",response="92cde606c2012aad89e95c8d1ddf0e77",algorithm=MD5,cnonce="abdcca769628eaf4225f8366058419d1",qop=auth,nc=00000001 Route: <sip:107.23.59.74:5060;lr;transport=udp> User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/ Organization: Doubango Telecom v=0 o=- 2443276864 2 IN IP4 127.0.0.1 s=Doubango Telecom - PeerConnection t=0 0 a=group:BUNDLE audio video m=audio 57473 RTP/SAVP 103 104 0 8 106 105 13 126 c=IN IP4 46.246.201.235 a=rtcp:57473 IN IP4 46.246.201.235 a=candidate:3802297132 1 udp 2113937151 192.168.0.3 57472 typ host generation 0 a=candidate:3802297132 2 udp 2113937151 192.168.0.3 57472 typ host generation 0 a=candidate:1274936569 1 udp 1677729535 46.246.201.235 57473 typ srflx generation 0 a=candidate:1274936569 2 udp 1677729535 46.246.201.235 57473 typ srflx generation 0 a=candidate:2887880668 1 tcp 1509957375 192.168.0.3 50736 typ host generation 0 a=candidate:2887880668 2 tcp 1509957375 192.168.0.3 50736 typ host generation 0 a=ice-ufrag:8tg3B+2oWiN3sm7T a=ice-pwd:3sqK8PM+sqnw4CXN1ba9nSB0 a=ice-options:google-ice a=sendrecv a=mid:audio a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:e8yvC9XIVcCUKtVYTDYlwkxPnQhvRtjzttMbDlCc a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=ssrc:187558285 cname:lzjjnqBxw4gUofQn a=ssrc:187558285 mslabel:0KwLFawpJwjTWNLWqVKYQtPrhuHnDMVbz6BF a=ssrc:187558285 label:0KwLFawpJwjTWNLWqVKYQtPrhuHnDMVbz6BF00 /src/tinySAK/src/tsk_utils.js?svn=13:59

__tsip_transport_ws_onmessage /src/tinySAK/src/tsk_utils.js?svn=13:59

recv=SIP/2.0 180 Ringing Via: SIP/2.0/TCP 46.246.201.235:50587;rport=50587;received=46.246.201.235;branch=z9hG4bKcK4Ws75sM3nZIqUmq3nK5XOHdPtRcFiN From: <sip:21303...@107.23.59.74>;tag=TIQtKygCbCAPoVtU7DpO To: <sip:9999...@107.23.59.74>;tag=cQrace98U97Qc Contact: <sip:9999...@10.0.1.59:5060;transport=udp> Call-ID: 7de16dc9-41b9-b1c6-b085-1b891489813e CSeq: 16406 INVITE Content-Length: 0 Accept: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,UPDATE,INFO,REGISTER,REFER,NOTIFY,PUBLISH,SUBSCRIBE Server: webrtc2sip Supported: timer,precondition,path,replaces User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120616T193227Z~068586f28f Allow-Events: talk,hold,presence,dialog,line-seize,call-info,sla,include-session-description,presence.winfo,message-summary,refer Remote-Party-ID: screen=no;party=calling;privacy=off;screen=no

On Wednesday, September 5, 2012 2:18:40 PM UTC+3, Anton wrote:

i've added in the end of
function tmedia_session_jsep.prototype.decorate_lo
in file
src\tinyMEDIA\src\tmedia_session_jsep.js

--
 
 

[Ros P]

Looks like you got some new content there. I will have a look, investigate and keep you posted.

thank you very much!! You and your team are doing great work ;)

--
 
 

[Ros P]

Still having a bit of trouble with media. I'm suspecting its due to the tcap media attribute in the SDP. I'm guessing it should be RTP/SAVP. Any ideas on how to change it?

v=0

o=doubango 1983 678901 IN IP4 10.0.1.204

s=-

c=IN IP4 10.0.1.204

t=0 0

m=audio 32802 RTP/AVP 0 8 101

c=IN IP4 10.0.1.204

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000/1

a=rtpmap:101 telephone-event/8000/1

a=fmtp:101 0-16

a=ptime:20

a=silenceSupp:off - - - -

a=tcap:1 RTP/SAVPF

a=pcfg:1 t=1

a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:hEAQVhUIvXsbX81X6EhUsYTS2WfepJ8OU8ElehL7

a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:mM9Ht4PdinZG2X/pSuXzkGBblvoAyvEW3JOkspaz

a=rtcp-mux

a=ssrc:4252906840 cname:ldjWoB60jbyQlR6e

a=ssrc:4252906840 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2

a=ssrc:4252906840 label:Doubango

a=ice-ufrag:ahUbjdxuyezUwlo

a=ice-pwd:MTPwRencw5v1G2W20DDG9

a=mid:audio

On Tuesday, December 4, 2012 12:58:17 AM UTC+2, Ros P wrote:

Looks like you got some new content there. I will have a look, investigate and keep you posted.

thank you very much!! You and your team are doing great work ;)

On Mon, Dec 3, 2012 at 7:26 PM, Mamadou DIOP <diopm...@doubango.org> wrote:

 http://code.google.com/p/webrtc2sip/#Testing_the_gateway

On Nov 30, 2012, at 9:58 PM, Ros P <rosph...@gmail.com> wrote:

Ive been able to successfully register a device using webrtc2sip gw, however i cant get any sound. The webserver, gw, and freeswitch are all on the same machine. Any ideas would be greatly appreciated?

State machine: x0000_Any_2_Any_X_i401_407_INVITE /src/tinySAK/src/tsk_utils.js?svn=13:59

SEND: INVITE sip:9999...@107.23.59.74 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcK4Ws75sM3nZIqUmq3nK5XOHdPtRcFiN;rport From: <sip:21303...@107.23.59.74>;tag=TIQtKygCbCAPoVtU7DpO To: <sip:9999...@107.23.59.74> Contact: "sp"<sip:2130334003@df7jal23ls0d.invalid;transport=ws>;+sip.ice Call-ID: 7de16dc9-41b9-b1c6-b085-1b891489813e CSeq: 16406 INVITE Content-Type: application/sdp Content-Length: 1294 Max-Forwards: 70 Proxy-Authorization: Digest username="2130334003",realm="107.23.59.74",nonce="550d599a-3b30-11e2-a729-8b18d47feaee",uri="sip:9999...@107.23.59.74",response="92cde606c2012aad89e95c8d1ddf0e77",algorithm=MD5,cnonce="abdcca769628eaf4225f8366058419d1",qop=auth,nc=00000001 Route: <sip:107.23.59.74:5060;lr;transport=udp> User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/ Organization: Doubango Telecom v=0 o=- 2443276864 2 IN IP4 127.0.0.1 s=Doubango Telecom - PeerConnection t=0 0 a=group:BUNDLE audio video m=audio 57473 RTP/SAVP 103 104 0 8 106 105 13 126 c=IN IP4 46.246.201.235 a=rtcp:57473 IN IP4 46.246.201.235 a=candidate:3802297132 1 udp 2113937151 192.168.0.3 57472 typ host generation 0 a=candidate:3802297132 2 udp 2113937151 192.168.0.3 57472 typ host generation 0 a=candidate:1274936569 1 udp 1677729535 46.246.201.235 57473 typ srflx generation 0 a=candidate:1274936569 2 udp 1677729535 46.246.201.235 57473 typ srflx generation 0 a=candidate:2887880668 1 tcp 1509957375 192.168.0.3 50736 typ host generation 0 a=candidate:2887880668 2 tcp 1509957375 192.168.0.3 50736 typ host generation 0 a=ice-ufrag:8tg3B+2oWiN3sm7T a=ice-pwd:3sqK8PM+sqnw4CXN1ba9nSB0 a=ice-options:google-ice a=sendrecv a=mid:audio a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:e8yvC9XIVcCUKtVYTDYlwkxPnQhvRtjzttMbDlCc a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=ssrc:187558285 cname:lzjjnqBxw4gUofQn a=ssrc:187558285 mslabel:0KwLFawpJwjTWNLWqVKYQtPrhuHnDMVbz6BF a=ssrc:187558285 label:0KwLFawpJwjTWNLWqVKYQtPrhuHnDMVbz6BF00 /src/tinySAK/src/tsk_utils.js?svn=13:59

--
 
 

--
 
 

[Mamadou]

Hard to help with without more info.
If you think that changing this could help: http://code.google.com/p/doubango/source/browse/branches/2.0/doubango/tinyDAV/src/tdav_session_av.c?r=776#526

--
 
 

[jwr99]

I ran into the exact same problem while trying to get webrtc2sip work with FreeSWITCH.  Hacking the doubango source as suggested to make the a-line "a=tcap:1 RTP/SAVP" did not help as shown in this FreeSWITCH trace.  Any ideas?  The SIP signaling is perfect, but FreeSWITCH returns 488 Not Acceptable Here based upon the SDP.

Thanks for your help earlier in building the webrtc2sip code and still very much looking forward to using it.  

- Joel

version

FreeSWITCH Version 1.3.8b+git~20121205T191750Z~924c524197 (git 924c524 2012-12-05 19:17:50Z)

. . . . 

   ------------------------------------------------------------------------

recv 1931 bytes from udp/[10.159.25.56]:10060 at 22:42:51.414083:

   ------------------------------------------------------------------------

   INVITE sip:9195@ . . . 

    . . . 

   User-Agent: webrtc2sip Media Server 2.0

   P-Preferred-Identity: <sip:webrtc2sip>

   v=0

   o=doubango 1983 678901 IN IP4 10.159.25.56

   s=-

   c=IN IP4 10.159.25.56

   t=0 0

   m=audio 60326 RTP/AVP 0 8 101

   c=IN IP4 10.159.25.56

   a=ptime:20

   a=silenceSupp:off - - - -

   a=rtpmap:0 PCMU/8000/1

   a=rtpmap:8 PCMA/8000/1

   a=rtpmap:101 telephone-event/8000/1

   a=fmtp:101 0-16

->   a=tcap:1 RTP/SAVP

   a=pcfg:1 t=1

   a=sendrecv

   a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:CG0ol5gUQjNxOXyMhXSIV2RltFltbx99IkjHmXsT

   a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:aiZ33ixxaTy5dX5iy72n/OAORs8lIezYKhuzti6G

   a=rtcp-mux

   a=ssrc:2783445458 cname:ldjWoB60jbyQlR6e

   a=ssrc:2783445458 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2

   a=ssrc:2783445458 label:Doubango

   a=ice-ufrag:kWutfVwWe7bWeB4

   a=ice-pwd:ziAr5WnKhzu1KsZsMinJw

   a=mid:audio

   a=candidate:35wALsfKp 1 udp 2130706431 10.159.25.56 60326 typ host

   a=candidate:35wALsfKp 2 udp 2130706430 10.159.25.56 60327 typ host

   a=candidate:srflx35wA 2 udp 1694498814 107.21.197.144 60327 typ srflx

   a=candidate:srflx35wA 1 udp 1694498815 107.21.197.144 60326 typ srflx

   ------------------------------------------------------------------------

send 400 bytes to udp/[10.159.25.56]:10060 at 22:42:51.414473:

   ------------------------------------------------------------------------

   SIP/2.0 100 Trying

   . . . 

   ------------------------------------------------------------------------

2012-12-11 22:42:51.734425 [INFO] mod_dialplan_xml.c:498 Processing 1001 <1001>->9195 in context default

-> 2012-12-11 22:42:51.894429 [ERR] sofia_glue.c:4922 a=crypto in RTP/AVP, refer to rfc3711

2012-12-11 22:42:51.894429 [NOTICE] switch_channel.c:3484 Hangup sofia/internal/1001@. . . 

 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]

send 924 bytes to udp/[10.159.25.56]:10060 at 22:42:52.054746:

   ------------------------------------------------------------------------

   SIP/2.0 488 Not Acceptable Here

   Via: SIP/2.0/UDP 10.159.25.56:10060;branch=z9hG4bK1356314285436;rport=10060

[Mamadou DIOP]

The SDP is correct.

You should ask on FreeSWITCH mailing list whet "[CS_EXECUTE] [INCOMPATIBLE_DESTINATION]" means. A quick search on google gives many results.

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[Mamadou DIOP]

I missed anther error message in your post: [ERR] sofia_glue.c:4922 a=crypto in RTP/AVP, refer to rfc3711

This is clearly a problem on FreeSWITCH (http://jira.freeswitch.org/browse/FS-636). They should not reject the stream based on these rules unless SRTP is mandatory.

It was also discussed on this list: https://groups.google.com/forum/#!msg/doubango/m7FCqYxRcQc/ajd_eC_oynYJ

To force using SAVP (not recommended), set the SRTP mode to mandatory: http://code.google.com/p/webrtc2sip/source/browse/trunk/mp_proxyplugin_mgr.cc?r=12#62

Setting SRTP mode in the config file will be allowed in the coming releases: http://code.google.com/p/webrtc2sip/issues/detail?id=33

On Dec 12, 2012, at 12:00 AM, jwr99 <joelrose...@gmail.com> wrote:

--
 
 

[Mamadou DIOP]

Looks like same config could be done on FreeSWITCH to fix the issue: http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_I_see_this_on_my_console_when_calling_FreeSWITCH_with_a_Snom_phone_.22a.3Dcrypto_in_RTP.2FAVP.2C_refer_to_RFC_3711.22_how_can_I_fix_this.3F

--
 
 

[Mamadou]

Fixed in webrtc2sip SVN r34: http://code.google.com/p/webrtc2sip/wiki/FAQ?ts=1355271529&updated=FAQ#I_see_"a=crypto_in_RTP/AVP,_refer_to_RFC_3711"_on_my_c

--
 
 

[jwr99]

That worked, thanks so much!  Well, it worked after I un-muted the speaker on my laptop.  :0

Specifically, I verified 2-way audio on Chrome version 23, webrtc2sip release 34, FreeSWITCH version 1.3.8

Set this webrtc2sip config.xml:   <srtp-mode>mandatory</srtp-mode> {I think this requires release 34}

With the default FS configuration, use sipml5 to Login (SIP REGISTER) to account 1001, then Call 9195 (SIP INVITE) to verify audio (5-second echo)

I'll continue to pursue the FreeSWITCH list to see if someone will answer the SDP question, but I have plenty to work with now.