Im using if(sdp_with_transport("RTP/SAVP")) to detect with endpoint is send SAVP or not to divert to and rtp proxy or rtpengine, as you know rtpproxy supports recording and rtpengine does not yet.
So when using if(sdp_with_transport("RTP/SAVP")) with Grandstream Phones all worked fine, but when configuring Optional or Compulsory SRTP in Yealink it seems to do not detect
i have seen that crypto lines are not in the final SDP but do not know if thats the reason
Did you have a similar issue with Yealink?
If i could get traces in anyway to help let me know.
BR
Alberto
INVITE sip:212@10.0.1.34:15060 SIP/2.0.
Record-Route: sip:x.x.x.x:8002;r2=on;lr=on;ftag=1072578853;nat=yes.
Record-Route: sip:x.x.x.x:8001;transport=tls;r2=on;lr=on;ftag=1072578853;nat=yes.
Via: SIP/2.0/UDP x.x.x.x.:8002;branch=z9hG4bK24c2.948e5074172530002b3bfb131ba51de6.0;i=1.
Via: SIP/2.0/TLS 10.0.1.111:11891 ;received=83.x.x.x;rport=11891;branch=z9hG4bK456460360.
From: "214" sip:214@1x.x.x.x:8001;tag=1072578853.
To: sip:212@x.x.x.x:8001.
Call-ID: 0_1310998066@10.0.1.111.
CSeq: 2 INVITE.
Contact: sip:214@83.x.x.x:11891;transport=TLS.
Content-Type: application/sdp.
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE.
Max-Forwards: 69.
User-Agent: Yealink SIP-T21P_E2 52.80.0.3.
Allow-Events: talk,hold,conference,refer,check-sync.
Content-Length: 553.
.
v=0.
o=- 20143 20143 IN IP4 x.x.x.x.
s=SDP data.
c=IN IP4 x.x.x.x
t=0 0.
m=audio 8530 RTP/AVP 0 8 18 101.
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:N2RjYzlhMjNmMzAwMDU5YzU2YjQ4ZTU1ODE4MzNm.
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NWQwYzgzMzhlYmU1OGY2NThmMzk2NjYwMTllZWI3.
a=crypto:3 F8_128_HMAC_SHA1_80 inline:YjEyN2M5Nzk4YzRmZDQ5ZTYxZGUzNTI3Yzg1YTgw.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.
a=nortpproxy:yes
Hello,
can you provide the sdp bodies for both Grandstream (that matched) and Yealink (that didn't match). We have to compare how the SAVP is advertised and how the function is making the check.
Cheers, Daniel
On 08/07/15 16:45, Alberto Sagredo wrote:
Im using if(sdp_with_transport("RTP/SAVP")) to detect with endpoint is send SAVP or not to divert to and rtp proxy or rtpengine, as you know rtpproxy supports recording and rtpengine does not yet.
So when using if(sdp_with_transport("RTP/SAVP")) with Grandstream Phones all worked fine, but when configuring Optional or Compulsory SRTP in Yealink it seems to do not detect
i have seen that crypto lines are not in the final SDP but do not know if thats the reason
Did you have a similar issue with Yealink?
If i could get traces in anyway to help let me know.
BR
Alberto
INVITE sip:212@10.0.1.34:15060 http://sip:212@10.0.1.34:15060 SIP/2.0.
Record-Route: sip:x.x.x.x:8002;r2=on;lr=on;ftag=1072578853;nat=yes.
Record-Route: sip:x.x.x.x:8001;transport=tls;r2=on;lr=on;ftag=1072578853;nat=yes.
Via: SIP/2.0/UDP x.x.x.x.:8002;branch=z9hG4bK24c2.948e5074172530002b3bfb131ba51de6.0;i=1.
Via: SIP/2.0/TLS 10.0.1.111:11891;received=83.x.x.x;rport=11891;branch=z9hG4bK456460360.
From: "214" sip:214@1x.x.x.x:8001;tag=1072578853.
To: sip:212@x.x.x.x:8001.
Call-ID: 0_1310998066@10.0.1.111 mailto:0_1310998066@10.0.1.111.
CSeq: 2 INVITE.
Contact: sip:214@83.x.x.x:11891;transport=TLS.
Content-Type: application/sdp.
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE.
Max-Forwards: 69.
User-Agent: Yealink SIP-T21P_E2 52.80.0.3.
Allow-Events: talk,hold,conference,refer,check-sync.
Content-Length: 553.
.
v=0.
o=- 20143 20143 IN IP4 x.x.x.x.
s=SDP data.
c=IN IP4 x.x.x.x
t=0 0.
m=audio 8530 RTP/AVP 0 8 18 101.
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:N2RjYzlhMjNmMzAwMDU5YzU2YjQ4ZTU1ODE4MzNm.
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NWQwYzgzMzhlYmU1OGY2NThmMzk2NjYwMTllZWI3.
a=crypto:3 F8_128_HMAC_SHA1_80 inline:YjEyN2M5Nzk4YzRmZDQ5ZTYxZGUzNTI3Yzg1YTgw.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.
a=nortpproxy:yes
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi Daniel
Here its Yealink one (Optional SRTP)
If you need anything more let me know
INVITE sip:1@192.168.0.181:5080 SIP/2.0.
Record-Route: <sip:x.x.x.x 8002;r2=on;lr=on;ftag=4139505128;nat=yes>.
Record-Route: sip:x.x.x.x:8001;transport=tls;r2=on;lr=on;ftag=4139505128;nat=yes.
Via: SIP/2.0/UDP x.x.x.x:8002;branch=z9hG4bK4928.64d27b89d3de27a54610b8ebb2aa9f43.0;i=2b2.
Via: SIP/2.0/TLS 10.0.1.111:11880 ;received=x.x.x.x;rport=11880;branch=z9hG4bK1819432518.
From: "214" sip:214@x.x.x.x:8001;tag=4139505128.
To: sip:1@x.x.x.x:8001.
Call-ID: 0_3807548115@10.0.1.111.
CSeq: 2 INVITE.
Contact: sip:214@80.x.x.x:11880;transport=TLS.
Content-Type: application/sdp.
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE.
Max-Forwards: 69.
User-Agent: Yealink SIP-T21P_E2 52.80.0.3.
Allow-Events: talk,hold,conference,refer,check-sync.
Content-Length: 549.
.
v=0.
o=- 20005 20005 IN IP4 192.168.0.178.
s=SDP data.
c=IN IP4 192.168.0.178.
t=0 0.
m=audio 8546 RTP/AVP 0 8 18 101.
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MjI1MDczY2JjYTM4MjM0MyBlMmIyZGI2YmUyZWI1.
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:N2EwZjhkMjAxMjlkMmFjMjcyY2E5NDczODM3Yjdh.
a=crypto:3 F8_128_HMAC_SHA1_80 inline:IDQ2YTBiYzQ2MDA1Y2ZhYWNkNTZmNmQ5NWY4Yjcw.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.
a=nortpproxy:yes.
And a GS one (Optional SRTP)
U 192.168.0.170:8002 -> 192.168.0.181:5080
INVITE sip:2@192.168.0.181:5080 SIP/2.0.
Record-Route: sip:x.x.x.x:8002;lr=on;ftag=429447500;nat=yes.
Via: SIP/2.0/UDP x.x.x.x:8002;branch=z9hG4bKc08.2bf157be2b7c1a44c1128d55db60357c.0.
Via: SIP/2.0/UDP x.x.x.x:46597;received=x.x.x.x;branch=z9hG4bK1529661043;rport=46597.
From: "Anonymous" sip:anonymous@anonymous.invalid;tag=429447500.
To: sip:2@x.x.x.x:8002.
Call-ID: 2055647556-46597-5@IA.CG.BIE.BCH.
CSeq: 40 INVITE.
Contact: "Anonymous" sip:212@x.x.x.x:46597.
X-Grandstream-PBX: true.
Max-Forwards: 69.
User-Agent: Grandstream GXP2140 1.0.4.23.
Privacy: id.
P-Preferred-Identity: sip:212@x.x.x.x:8002.
Supported: replaces, path, timer.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE.
Content-Type: application/sdp.
Accept: application/sdp, application/dtmf-relay.
Content-Length: 753.
.
v=0.
o=212 8000 8000 IN IP4 x.x.x.x
s=SIP Call.
c=IN IP4 x.x.x.x.
t=0 0.
m=audio 32584 RTP/AVP 0 8 18 9 2 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:9 G722/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
m=audio 32584 RTP/SAVP 0 8 18 9 2 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:9 G722/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/cVB/SqgmIibo+CJTVZvnDNOf9dNxFFaQc70pqbm.
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:76OrMKDV0Dhda9w+9SmUZMbHskWe/wnwWUq+TfFk.
2015-07-13 9:19 GMT+02:00 Daniel-Constantin Mierla miconda@gmail.com:
Hello,
can you provide the sdp bodies for both Grandstream (that matched) and Yealink (that didn't match). We have to compare how the SAVP is advertised and how the function is making the check.
Cheers, Daniel
On 08/07/15 16:45, Alberto Sagredo wrote:
Im using if(sdp_with_transport("RTP/SAVP")) to detect with endpoint is send SAVP or not to divert to and rtp proxy or rtpengine, as you know rtpproxy supports recording and rtpengine does not yet.
So when using if(sdp_with_transport("RTP/SAVP")) with Grandstream Phones all worked fine, but when configuring Optional or Compulsory SRTP in Yealink it seems to do not detect
i have seen that crypto lines are not in the final SDP but do not know if thats the reason
Did you have a similar issue with Yealink?
If i could get traces in anyway to help let me know.
BR
Alberto
INVITE sip:212@10.0.1.34:15060 SIP/2.0.
Record-Route: sip:x.x.x.x:8002;r2=on;lr=on;ftag=1072578853;nat=yes.
Record-Route: sip:x.x.x.x:8001;transport=tls;r2=on;lr=on;ftag=1072578853;nat=yes.
Via: SIP/2.0/UDP x.x.x.x.:8002;branch=z9hG4bK24c2.948e5074172530002b3bfb131ba51de6.0;i=1.
Via: SIP/2.0/TLS 10.0.1.111:11891 ;received=83.x.x.x;rport=11891;branch=z9hG4bK456460360.
From: "214" sip:214@1x.x.x.x:8001;tag=1072578853.
To: sip:212@x.x.x.x:8001.
Call-ID: 0_1310998066@10.0.1.111.
CSeq: 2 INVITE.
Contact: sip:214@83.x.x.x:11891;transport=TLS.
Content-Type: application/sdp.
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE.
Max-Forwards: 69.
User-Agent: Yealink SIP-T21P_E2 52.80.0.3.
Allow-Events: talk,hold,conference,refer,check-sync.
Content-Length: 553.
.
v=0.
o=- 20143 20143 IN IP4 x.x.x.x.
s=SDP data.
c=IN IP4 x.x.x.x
t=0 0.
m=audio 8530 RTP/AVP 0 8 18 101.
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:N2RjYzlhMjNmMzAwMDU5YzU2YjQ4ZTU1ODE4MzNm.
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NWQwYzgzMzhlYmU1OGY2NThmMzk2NjYwMTllZWI3.
a=crypto:3 F8_128_HMAC_SHA1_80 inline:YjEyN2M5Nzk4YzRmZDQ5ZTYxZGUzNTI3Yzg1YTgw.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.
a=nortpproxy:yes
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Alberto Sagredo -
Daniel-Constantin Mierla