RTPProxy nortpproxy_str issue

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RTPProxy nortpproxy_str issue

Seth Schultz
Hello,

I am having a problem with RTPProxy where in the reply, the remote
carrier is sending the "nortpproxy_str" in the reply SDP (example
below).  I would like to know what the best way is to detect this, and
remove it from the sip message before calling rtpproxy_answer function,
because rtpproxy_answer will fail if the nortpproxy_str already exists
in the SDP.

Thanks in advance,
Seth

U 2013/02/14 19:32:21.142567 yyy.yyy.yyy.yyy:5060 -> xxx.xxx.xxx.xxx:5060
INVITE sip:[hidden email] SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2d9e.187ebf5.0
Max-Forwards: 69
From: "Unknown" <sip:[hidden email]>;tag=33XjNy6SQZrQS
To: <sip:[hidden email]>
Call-ID: 004c5840-f1aa-1230-9c93-6320dec8e883
CSeq: 40108106 INVITE
Contact: <sip:yyy.yyy.yyy.yyy;did=3901.59b3bb21>
User-Agent: FS1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 247
P-Call-Type: Notification
X-FS-Support: update_display,send_info
Remote-Party-ID: "Unknown"
<sip:[hidden email]>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1360855702 1360855703 IN IP4 yyy.yyy.yyy.yyy
s=FreeSWITCH
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 40562 RTP/AVP 0 8 3 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=schipmangled:yes  <--- rtpproxy added this on initial invite

...

U 2013/02/14 19:32:37.425606 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2d9e.187ebf5.0
From: "Unknown" <sip:[hidden email]>;tag=33XjNy6SQZrQS
To: <sip:[hidden email]>;tag=SDs07f299-gK0e9f2e8d
Call-ID: 004c5840-f1aa-1230-9c93-6320dec8e883
CSeq: 40108106 INVITE
Accept: application/sdp, application/isup, application/dtmf,
application/dtmf-relay,  multipart/mixed
Contact: <sip:xxx.xxx.xxx.xxx;did=39.60d51ef>
Allow:
INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
Require: timer
Supported: timer
Session-Expires: 7200;refresher=uas
Content-Length: 259
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 7607 20874 IN IP4 xxx.xxx.xxx.xxx
s=SIP Media Capabilities
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 29772 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=schipmangled:yes  <--- they sent this back in the 200 OK reply
a=ptime:20
a=sendrecv


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Users mailing list
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Re: RTPProxy nortpproxy_str issue

shaheryarkh
You mean both you and your carrier are using their own rtp-proxy? If so, then simply add "f" flag to rtpproxy_offer | rtpproxy_answer. Which will allow you can you carrier to create a chain of rtp-proxy together. See flags description here,

http://www.opensips.org/html/docs/modules/devel/rtpproxy.html#id292744

Thank you.


On Fri, Feb 15, 2013 at 2:18 AM, Seth Schultz <[hidden email]> wrote:
Hello,

I am having a problem with RTPProxy where in the reply, the remote carrier is sending the "nortpproxy_str" in the reply SDP (example below).  I would like to know what the best way is to detect this, and remove it from the sip message before calling rtpproxy_answer function, because rtpproxy_answer will fail if the nortpproxy_str already exists in the SDP.

Thanks in advance,
Seth

U 2013/02/14 19:32:21.142567 yyy.yyy.yyy.yyy:5060 -> xxx.xxx.xxx.xxx:5060
INVITE sip:[hidden email].xxx SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2d9e.187ebf5.0
Max-Forwards: 69
From: "Unknown" <sip:[hidden email].yyy>;tag=33XjNy6SQZrQS
To: <sip:[hidden email].yyy>
Call-ID: 004c5840-f1aa-1230-9c93-6320dec8e883
CSeq: 40108106 INVITE
Contact: <sip:yyy.yyy.yyy.yyy;did=3901.59b3bb21>
User-Agent: FS1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 247
P-Call-Type: Notification
X-FS-Support: update_display,send_info
Remote-Party-ID: "Unknown" <sip:[hidden email].yyy>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1360855702 1360855703 IN IP4 yyy.yyy.yyy.yyy
s=FreeSWITCH
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 40562 RTP/AVP 0 8 3 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=schipmangled:yes  <--- rtpproxy added this on initial invite

...

U 2013/02/14 19:32:37.425606 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2d9e.187ebf5.0
From: "Unknown" <sip:[hidden email].yyy>;tag=33XjNy6SQZrQS
To: <sip:[hidden email].yyy>;tag=SDs07f299-gK0e9f2e8d
Call-ID: 004c5840-f1aa-1230-9c93-6320dec8e883
CSeq: 40108106 INVITE
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay,  multipart/mixed
Contact: <sip:xxx.xxx.xxx.xxx;did=39.60d51ef>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
Require: timer
Supported: timer
Session-Expires: 7200;refresher=uas
Content-Length: 259
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 7607 20874 IN IP4 xxx.xxx.xxx.xxx
s=SIP Media Capabilities
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 29772 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=schipmangled:yes  <--- they sent this back in the 200 OK reply
a=ptime:20
a=sendrecv


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



--
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: [hidden email]
Email: [hidden email]
_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: RTPProxy nortpproxy_str issue

Seth Schultz
Muhammad,

I don't know what the remote carrier is using for their RTP.  I set a custom nortpproxy_str to try and avoid this (instead of leaving it as the default a=nortpproxy:yes).  Is it correct for them to leave our custom a=schipmangled:yes record in the SDP?  I have had problems with the "f" flag and failover routing (basically rewrites the IP in the SDP twice like this yyy.yyy.yyy.yyyyyy.yyy.yyy.yyy).  Is there an easy way for me to just remove the a=schipmangle:yes in my onreply_route?

Thanks,
Seth

On 2/14/2013 8:28 PM, Muhammad Shahzad wrote:
You mean both you and your carrier are using their own rtp-proxy? If so, then simply add "f" flag to rtpproxy_offer | rtpproxy_answer. Which will allow you can you carrier to create a chain of rtp-proxy together. See flags description here,

http://www.opensips.org/html/docs/modules/devel/rtpproxy.html#id292744

Thank you.


On Fri, Feb 15, 2013 at 2:18 AM, Seth Schultz <[hidden email]> wrote:
Hello,

I am having a problem with RTPProxy where in the reply, the remote carrier is sending the "nortpproxy_str" in the reply SDP (example below).  I would like to know what the best way is to detect this, and remove it from the sip message before calling rtpproxy_answer function, because rtpproxy_answer will fail if the nortpproxy_str already exists in the SDP.

Thanks in advance,
Seth

U 2013/02/14 19:32:21.142567 yyy.yyy.yyy.yyy:5060 -> xxx.xxx.xxx.xxx:5060
INVITE [hidden email] SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2d9e.187ebf5.0
Max-Forwards: 69
From: "Unknown" [hidden email];tag=33XjNy6SQZrQS
To: [hidden email]
Call-ID: 004c5840-f1aa-1230-9c93-6320dec8e883
CSeq: 40108106 INVITE
Contact: <sip:yyy.yyy.yyy.yyy;did=3901.59b3bb21>
User-Agent: FS1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 247
P-Call-Type: Notification
X-FS-Support: update_display,send_info
Remote-Party-ID: "Unknown" [hidden email];party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1360855702 1360855703 IN IP4 yyy.yyy.yyy.yyy
s=FreeSWITCH
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 40562 RTP/AVP 0 8 3 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=schipmangled:yes  <--- rtpproxy added this on initial invite

...

U 2013/02/14 19:32:37.425606 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2d9e.187ebf5.0
From: "Unknown" [hidden email];tag=33XjNy6SQZrQS
To: [hidden email];tag=SDs07f299-gK0e9f2e8d
Call-ID: 004c5840-f1aa-1230-9c93-6320dec8e883
CSeq: 40108106 INVITE
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay,  multipart/mixed
Contact: <sip:xxx.xxx.xxx.xxx;did=39.60d51ef>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
Require: timer
Supported: timer
Session-Expires: 7200;refresher=uas
Content-Length: 259
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 7607 20874 IN IP4 xxx.xxx.xxx.xxx
s=SIP Media Capabilities
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 29772 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=schipmangled:yes  <--- they sent this back in the 200 OK reply
a=ptime:20
a=sendrecv


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



--
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: [hidden email]
Email: [hidden email]

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Reply | Threaded
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Re: RTPProxy nortpproxy_str issue

shaheryarkh
Yes, you can use this method,


e.g. something like,

if (has_body("application/sdp") && replace_body_atonce("a=schipmangle:yes", ""))
   xlog("Removed a=schipmangle:yes from carrier xxx");

Thank you.


On Fri, Feb 15, 2013 at 2:53 AM, Seth Schultz <[hidden email]> wrote:
Muhammad,

I don't know what the remote carrier is using for their RTP.  I set a custom nortpproxy_str to try and avoid this (instead of leaving it as the default a=nortpproxy:yes).  Is it correct for them to leave our custom a=schipmangled:yes record in the SDP?  I have had problems with the "f" flag and failover routing (basically rewrites the IP in the SDP twice like this yyy.yyy.yyy.yyyyyy.yyy.yyy.yyy).  Is there an easy way for me to just remove the a=schipmangle:yes in my onreply_route?

Thanks,
Seth


On 2/14/2013 8:28 PM, Muhammad Shahzad wrote:
You mean both you and your carrier are using their own rtp-proxy? If so, then simply add "f" flag to rtpproxy_offer | rtpproxy_answer. Which will allow you can you carrier to create a chain of rtp-proxy together. See flags description here,

http://www.opensips.org/html/docs/modules/devel/rtpproxy.html#id292744

Thank you.


On Fri, Feb 15, 2013 at 2:18 AM, Seth Schultz <[hidden email]> wrote:
Hello,

I am having a problem with RTPProxy where in the reply, the remote carrier is sending the "nortpproxy_str" in the reply SDP (example below).  I would like to know what the best way is to detect this, and remove it from the sip message before calling rtpproxy_answer function, because rtpproxy_answer will fail if the nortpproxy_str already exists in the SDP.

Thanks in advance,
Seth

U 2013/02/14 19:32:21.142567 yyy.yyy.yyy.yyy:5060 -> xxx.xxx.xxx.xxx:5060
INVITE [hidden email] SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2d9e.187ebf5.0
Max-Forwards: 69
From: "Unknown" [hidden email];tag=33XjNy6SQZrQS
To: [hidden email]
Call-ID: 004c5840-f1aa-1230-9c93-6320dec8e883
CSeq: 40108106 INVITE
Contact: <sip:yyy.yyy.yyy.yyy;did=3901.59b3bb21>
User-Agent: FS1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 247
P-Call-Type: Notification
X-FS-Support: update_display,send_info
Remote-Party-ID: "Unknown" [hidden email];party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1360855702 1360855703 IN IP4 yyy.yyy.yyy.yyy
s=FreeSWITCH
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 40562 RTP/AVP 0 8 3 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=schipmangled:yes  <--- rtpproxy added this on initial invite

...

U 2013/02/14 19:32:37.425606 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2d9e.187ebf5.0
From: "Unknown" [hidden email];tag=33XjNy6SQZrQS
To: [hidden email];tag=SDs07f299-gK0e9f2e8d
Call-ID: 004c5840-f1aa-1230-9c93-6320dec8e883
CSeq: 40108106 INVITE
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay,  multipart/mixed
Contact: <sip:xxx.xxx.xxx.xxx;did=39.60d51ef>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
Require: timer
Supported: timer
Session-Expires: 7200;refresher=uas
Content-Length: 259
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 7607 20874 IN IP4 xxx.xxx.xxx.xxx
s=SIP Media Capabilities
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 29772 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=schipmangled:yes  <--- they sent this back in the 200 OK reply
a=ptime:20
a=sendrecv


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



--
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: [hidden email]
Email: [hidden email]

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users




--
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: [hidden email]
Email: [hidden email]

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Reply | Threaded
Open this post in threaded view
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Re: RTPProxy nortpproxy_str issue

Seth Schultz
Muhammad,

Thanks... I will try this.

Seth

      On 2/15/2013 2:20 AM, Muhammad Shahzad wrote:
Yes, you can use this method,


e.g. something like,

if (has_body("application/sdp") && replace_body_atonce("a=schipmangle:yes", ""))
   xlog("Removed a=schipmangle:yes from carrier xxx");

Thank you.


On Fri, Feb 15, 2013 at 2:53 AM, Seth Schultz <[hidden email]> wrote:
Muhammad,

I don't know what the remote carrier is using for their RTP.  I set a custom nortpproxy_str to try and avoid this (instead of leaving it as the default a=nortpproxy:yes).  Is it correct for them to leave our custom a=schipmangled:yes record in the SDP?  I have had problems with the "f" flag and failover routing (basically rewrites the IP in the SDP twice like this yyy.yyy.yyy.yyyyyy.yyy.yyy.yyy).  Is there an easy way for me to just remove the a=schipmangle:yes in my onreply_route?

Thanks,
Seth


On 2/14/2013 8:28 PM, Muhammad Shahzad wrote:
You mean both you and your carrier are using their own rtp-proxy? If so, then simply add "f" flag to rtpproxy_offer | rtpproxy_answer. Which will allow you can you carrier to create a chain of rtp-proxy together. See flags description here,

http://www.opensips.org/html/docs/modules/devel/rtpproxy.html#id292744

Thank you.


On Fri, Feb 15, 2013 at 2:18 AM, Seth Schultz <[hidden email]> wrote:
Hello,

I am having a problem with RTPProxy where in the reply, the remote carrier is sending the "nortpproxy_str" in the reply SDP (example below).  I would like to know what the best way is to detect this, and remove it from the sip message before calling rtpproxy_answer function, because rtpproxy_answer will fail if the nortpproxy_str already exists in the SDP.

Thanks in advance,
Seth

U 2013/02/14 19:32:21.142567 yyy.yyy.yyy.yyy:5060 -> xxx.xxx.xxx.xxx:5060
INVITE [hidden email] SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2d9e.187ebf5.0
Max-Forwards: 69
From: "Unknown" [hidden email];tag=33XjNy6SQZrQS
To: [hidden email]
Call-ID: 004c5840-f1aa-1230-9c93-6320dec8e883
CSeq: 40108106 INVITE
Contact: <sip:yyy.yyy.yyy.yyy;did=3901.59b3bb21>
User-Agent: FS1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 247
P-Call-Type: Notification
X-FS-Support: update_display,send_info
Remote-Party-ID: "Unknown" [hidden email];party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1360855702 1360855703 IN IP4 yyy.yyy.yyy.yyy
s=FreeSWITCH
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 40562 RTP/AVP 0 8 3 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=schipmangled:yes  <--- rtpproxy added this on initial invite

...

U 2013/02/14 19:32:37.425606 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2d9e.187ebf5.0
From: "Unknown" [hidden email];tag=33XjNy6SQZrQS
To: [hidden email];tag=SDs07f299-gK0e9f2e8d
Call-ID: 004c5840-f1aa-1230-9c93-6320dec8e883
CSeq: 40108106 INVITE
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay,  multipart/mixed
Contact: <sip:xxx.xxx.xxx.xxx;did=39.60d51ef>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
Require: timer
Supported: timer
Session-Expires: 7200;refresher=uas
Content-Length: 259
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 7607 20874 IN IP4 xxx.xxx.xxx.xxx
s=SIP Media Capabilities
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 29772 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=schipmangled:yes  <--- they sent this back in the 200 OK reply
a=ptime:20
a=sendrecv


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



--
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: [hidden email]
Email: [hidden email]

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users




--
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: [hidden email]
Email: [hidden email]


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users